Networking between VOIP -and PSTN- calls

ABSTRACT

Programmatically reversing numerical line identity presented at a communications services gateway into named IP Telephony users with “prior association”, delivers dynamic “reverse address resolution” switching connections from ground to cloud, permitting any conventional telephone to dial and connect to any associated IP Telephony endpoint in the world, without changes to the conventional telephone. Reversing line identity into associated named users bridges both the addressability and economic divide between mass conventional “paying” (mobile and fixed) and “free”. IP Telephony networks. A system for supporting communications between a user on an IP-addressed-communications-device and a telephony subscriber device, the telephony subscriber device having a corresponding telephone number, includes: one or more service nodes configured to: receive from the user the telephone number of the telephony subscriber device and create an association from the telephone number to the user, wherein the association allows the telephony subscriber device to connect to the user.

TECHNICAL FIELD

This disclosure relates to telephony services on a network, and moreparticularly to Fixed, Mobile and IP telecommunications interworking.

BACKGROUND

Conventional Telephone Systems identify and address users with decimalnumbers geographically mapped on the E164 dial plan. As such thearchetypal telephone equipment and interface presents a numeric dialpad. Next Generation Networks, exemplified by Voice Over InternetProtocols (VOIP) register and identify its end users “by name ratherthan number”.

Conventional Telephones typically cannot dial the user address of anInternet Client using known methods. While Internet users can easilyenter phone numbers and thus readily address conventional telephones,the caveat in calling “offnet” to Fixed and Mobile telephones, is thatcalling “off net” often incurs a terminating penalty. That is, carrierson the ground charge to terminate voice and data traffic onto theirnetworks.

This carrier interoperating business rule forces VOIP service providersto charge its users for calling “off net”, and the reason theoverwhelming majority of VOIP calls remain on net. However while “freespeech” is the mass VOIP consumer proposition and draw card, calling “onnet” keeps communication “up in the cloud” and consequently fails togenerate the revenue required to commercialize service.

SUMMARY

The challenges facing the VOIP industry are as much related to realworld economics as they are to the culture that permeates on lineservices. Built on “the economics of free” the mass inertia to paying ispractically impossible to overcome. Notwithstanding the “free for all”,VOIP service providers are being forced out of economic necessity, to“peddle cheap”, discounting conventional phone service, and in doing so,tarnishing and eroding their brand, their vision and their serviceproposition.

Charging pennies, and now fractional pennies, for calls to legacytelephones around the world in order to survive, has become a fiercelycompetitive business. However, with less than 10 percent of allconnections resulting in off net billable traffic, the VOIP Industryremains “up in the cloud and vapor”.

While the Internet scrambles to capture the mass mobile paying market,by developing specialized applications that require smart phones andusers to “download and register” in order to gain access to onlineservices, the overwhelming majority lacks the means, the know how andthe device capability to do so.

Conversely, the Mobile industry has invested an untold fortune overdecades, pushing mobile data service adoption uphill from “4 to 10%”,relentlessly driven by the “StarTrek Tricorder” celluloid fantasy. Thecellular reality however, remains one in which some 80% of globaloperator revenue still accrues from people dialing telephone numbers andtalking. Technology changes at the speed of light, human behavior, “atsnail's pace”.

Ever faster, more capable and more data hungry devices now hold carriersin “golden data handcuffs”, disintermediating core voice service andconsuming vast amounts of wireless spectrum and network capacity thatdemand immense and continual infrastructure upgrades. With a constantlyshifting event horizon, data business models remain unstable andunproven. The “faster horse” mentality may get networks to an unintendeddestination quicker:

“More users disappointed with performance more often”.

Principally for these reasons, Convergence between core IP and MobileTelephony has become a 21st Century Telecom Mantra. However withoutfully grasping the mass behavioral phenomena that govern serviceadoption, “converging technology without converging culture” can be amassive miscalculation.

Whereas the Internet hallmark is “free”, mobile has “paying” as theestablished core proposition. These extreme, diametrically opposedecosystems create a massive potential and desire to interconnect withoutchanging user expectations and economic propositions on either.

Annotating now the deeper underlying technical, behavioral, cultural andeconomic drivers presently shaping the VOIP industry, with reference tothe Appended FIGS. 34 through 38.

FIG. 34

Requiring users to “Download, Register and Pay”, in short order presentsthe three inhibitors to mass service adoption. Notably Internet“search”, the most storied success on the web, foregoes all three. Whilemany broadband Internet users clear the first two hurdles with ease, thethird bar is way too high for a community built on free.

In the VOIP context, while hundreds of millions of users have downloadedand registered “10 Megabyte clients on their desktops”, less than “1percent” have offered up their “VISA cards online to pay for calls”.Clearly there is more to net success than gaining mass.

Things are a dramatically different in the cellular world. The mirroredreality here, is while the cellular masses are required to “jump and payup front” (global service provision model being prepaid) most hesitatewhen it comes to “downloading and registration”. Registering is “theprice of free”, since personal data is required for effectiveadvertising. However mass markets are notoriously fickle. Many valueanonymity.

The “Net Versus Cell” graph in FIG. 34 depicts these invertedmegatrends.

The mainstay of Internet Commercialization is third party advertisingand sponsorships. However these business models cannot be grafted from“visual to audio”. That is while “search, text and banner” adverts haveproven to be a sustainable business model on the Net given their “visualproximity to the supported service”, audio services engage users on adifferent plane.

Whereas it is viable to show collateral advertising and sponsorship inthe “field of vision”, engaging a user in a similar manner while“talking and listening” is significantly more challenging. This is inpart due to the fact that the “consumption and the carrot” (the servicebeing offered free in lieu of the advertising) are disassociated. Thatis the service is “consumed” independently of and irrespective to thesponsorship.

Furthermore, talking and listening being the natural human form ofcommunication is “emotionally charged”. Associating brands, products andservices with what is essentially an “unscripted production” can haveunintended and less than desirable consequence. For example, presentingproduct C on a call that results in a breakup between A and B can resultin a “lifetime of negative brand association”. This especially the casewhen the sponsoring message is presented “in ear, on call”.

Sponsored Telephony Services predicated on delivering “out of band”advertising, for example via text and multimedia messages sent eitherfollowing or preceding conversations, have not reached their promisedpotential. While these services have reported high “click through” onthe advertising channel “conversion” remains problematic (“click delete”is the more common consequence).

FIG. 36 illustrates different origination and termination callingpatterns between VOIP and Legacy networks and the resultant interconnectpayment. Calls between two IP users are “free on net”. Calls from IP tolegacy incur cost (IP outgoing interconnect fees, top circles).

Calls from Legacy to IP generate revenue for the Gateway provider (IPinbound interconnect, middle and lower circles) and those that “meet atthe edges” (IP inbound interconnect, lower circles) may be switched andbridged.

The conventional VOIP service caveat is “desktop in a mobile world”,requiring both users to be online available and connected at the sametime in order to talk free. However given the global service reachtogether with planetary “day/night” time zones and an increasinglymobile society, one user is typically at a fixed location (on net athome/office) while the other is mobile on cell.

Moreover, while IP to Legacy hits the proverbial billing brick wall,additional social factors come in to play when calling off net.Principal amongst these is the fact that telephones lack “presence”.While “always on and always connected”, there is still no known,acceptable and interoperable telephone presence protocol.

Paradoxically, Caller Line Identity, the very essence of moderntelephony, delivers at best “a reverse presence indication”, one thatindicates to the called party that the “caller is available and wishingto connect”. CLI is a double edged sword that undermines callcompletion, as many users now screen callers.

Playing telephone tag is, however, symptomatic of a much deeperunderlying design flaw being the synchronous and unilateral nature oftelecommunications, which permits anyone to “push and enter a ringuninvited”. The result is a technology, which sees less than “1 in 3mobile calls going to successful completion”.

Further, many of the calls that do successfully complete are themselvesshort lived (“I am busy, call you right back”) and the cost associatedwith switching, paging and wireless spectrum consumption for shortduration calls is prohibitive.

FIG. 37

“Free is one market, anything else is another”.

The mistaken impression is that there is constant elasticity in price,which the cheaper a service costs the more users it will attract.However the greatest gap in service is between “one that is free and onethat costs a penny”. The Penny Gap is the reverse salient that placessevere drag on the VOIP industry. The quantum is almost irrelevant. Themass net psychology has to do with “payment”.

FIG. 37 graphically illustrates these two phenomena. With respect toprice elasticity, the dotted band stretches to the “theoretical linearuptake”, whereas the solid band depicts the “actual limitation”. Thevertical cross section reveals a “submerged pyramid” beneath theseemingly small penny gap.

The “very known and met need” is in providing ever cheaper “almost free”ways to interconnect. There is therefore a need for the “unknown andunmet need” to programmatically switch and terminate conventional callsinto the cloud, and thereby deliver “a reverse free global connection”to the community on the net.

Reference will now be made in detail to embodiments, examples of whichare illustrated in the accompanying drawings. In the following detaileddescription, numerous specific details are set forth in order to providea sufficient understanding of the subject matter presented herein. Butit will be apparent to one of ordinary skill in the art that the subjectmatter may be practiced without these specific details. Moreover, theparticular embodiments described herein are provided by way of exampleand should not be used to limit the scope of the invention to theseparticular embodiments. In other instances, well-known data structures,timing protocols, software operations, procedures, and components havenot been described in detail so as not to unnecessarily obscure aspectsof the embodiments of the invention.

While the embodiments highlight voice telephony services, select reverseassociated addressing schemas and the methods and systems herein thatdescribe them, may be readily applied to alternate and complimentarycommunication protocols and services, including text, picture, video andmultimedia messaging.

FIG. 35

Since it is difficult to overcome “the net resistance to paying” andgiven that telephony plus advertising may be impractical, the onlyviable and sustainable mass VOIP economic proposition is in delivery andadoption on mobile, where “pay is in the DNA”.

To converge the known systems:net=+download+register−paycell=−download−register+paynet and cell=“zero sum” (game)

However as illustrated above, “goosing mobile” is contingent on removing“download and registration” from the cell equation. That is, foregoingthe downloading and installation of special VOIP software on select andsupporting devices. “No download and no registration” equates to“service virtualization”.

Restating the virtual equation:net=+download+registercell=+paynet and cell=“+++” (triple win)

Thus in order to deliver seamless and instant, local to global (groundto cloud) connectivity to any phone without change, the service isrequired to be “resident in the cloud” and accessed “over the air”, overthe ubiquitous conventionally established voice channel.

FIG. 36

Characterizing the VOIP problem in meteorological terms, terminating IPconnections onto Fixed and Cellular networks is “lightening that strikesdown and out”, discharging energy from “cloud to ground”, and incurringthe terminating penalty.

However, in Cellular Communication Systems there is a lesser knownelectrical phenomenon called “Ground to Cloud strikes”, where lighteningstrikes “up” from cell towers on the ground, “spectacularly lighting upthe cloud”.

The methods and systems disclosed similarly “lights up the VOIP cloud”and industry by uniquely switching legacy originated calls between“Ground and Cloud”. These “reversed electrical circuits” are closedbetween the “positive (+) revenue generating cells” and “negative (−)cloud atmospherics”.

FIG. 38

While at first the challenge to converge and uniquely map millions ofInternet users to billions of conventional telephones appearsunattainable, when reduced to practice the “social geometry” appears tobe inversely parabolic rather than exponential, linear or quadratic.That is, at the convergence of these two planetary telephony ecosystems,a mere handful of interconnections actually exist.

The “90:10” rule here is “over ninety percent only ever talk to tenindividuals or less” on a regular basis. This small data set makescontact presentation and selection exceedingly manageable, especiallygiven that the disclosed association between the two domains is“discoverable”. FIG. 38 (F38) plots this social geometry.

By reversing call line identity presented on the Ground into the Cloud,the methods and systems disclosed permit legacy phones to dial andconnect to any associated VOIP endpoint in the world without having toknow or manually enter the associated Internet User Identity. Thisextremely subtle yet profound advance in IP switching, transforms theVOIP service landscape.

Since making telephone calls remains the primary and consequently themost refined mobile phone function and user experience, it is useful toleverage this communications channel to deliver a service andexperience.

In both preserving the status quo in the ecosystems and reversing “cloudeconomics” as described, the methods and systems disclosed uniquelycaptures and converges conventional Telephony Markets, where callersroutinely pay for locally switched connections, with Internet TelephonyCommunities, who are set on connecting globally and speaking free.

Significantly, the service galvanizes voice by leveraging existingassets and infrastructure. Since legacy connections are circuitswitched, they clock minutes of use, locking in conventional callrevenue generation. Whereas before VOIP service providers and users hadto pay to connect to legacy telephones, they now receive connectionsgenerating net positive inbound revenues that are accrued to the VOIPservice provider on existing Fixed and Mobile Interconnect.

According to the teachings of the present embodiment there is provided asystem for supporting communications between an internet user on aninternet-addressed-communications-device and a telephony subscriberdevice, the telephony subscriber device having a corresponding telephonenumber, the system including: one or more service nodes configured to:receive from the internet-addressed-communications-device of theinternet user the telephone number of the telephony subscriber device;and create an association from the telephone number to the internetuser, wherein the association allows the telephony subscriber device toconnect to the internet user.

In an optional embodiment, at least one of the service nodes isconfigured to receive the telephone number of the telephony subscriberdevice in response to an action by the internet user.

In an optional embodiment, the system includes at least one gatewayconfigured to: receive a communications request from the telephonysubscriber device, the communications request including the telephonenumber of the telephony subscriber device; and cause, by using thereceived telephone number, retrieval of addressing information for theinternet user that is associated with the telephone number, wherein thesystem is configured to use the retrieved addressing information tocause establishment of a communications connection between the telephonysubscriber device and a device then being used by the internet user,thereby supporting communications between the telephony subscriberdevice and the internet user.

In an optional embodiment, the addressing information for the internetuser includes but is not limited to: an Internet service user address,an Internet service user name, an Internet URI, an IP address, an IPcommunications endpoint address, and a MAC address.

In an optional embodiment, in response to receiving the communicationsrequest from the telephony subscriber device, the system is furtherconfigured to instantiate a client proxy for the telephone number of thetelephony subscriber device, the client proxy operative to facilitatecommunications between the telephony subscriber device and the internetuser.

In an optional embodiment, at least one of the gateways is furtherconfigured to: receive information regarding a plurality of internetusers for which addressing information is associated with the telephonenumber, and configure, based on the information received, the clientproxy to facilitate communications between the telephony subscriberdevice and at least one of the internet users for which addressinginformation has been received.

In an optional embodiment, the client proxy includes status of thetelephony subscriber device. In another optional embodiment, the clientproxy is configured to publish information for the telephony subscriberdevice and the associated internet users. In another optionalembodiment, the client proxy is configured to translate betweencommunication protocols used respectively by the telephony subscriberdevice and one or more devices used by the internet user.

In an optional embodiment, the one or more service nodes is furtherconfigured to cause associations between the telephone number andaddressing information, the addressing information respectivelyassociated with a plurality of internet users; and the system is furtherconfigured to permit the telephony subscriber device to establishcommunications with at least one of the plurality of internet users.

In an optional embodiment, if addressing information of a singleavailable internet user is associated with the telephone number, atleast one gateway is configured to: notify the telephony subscriberdevice of a communications possibility; cause establishment of aconnection from the telephony subscriber device to the internet user;and cause the identification of the telephony subscriber device, as acalling party, to the internet user.

In an optional embodiment, at least one service gateway is configured,if a plurality of available internet users is associated with thetelephone number, to present to the telephony subscriber device a listof the plurality of available internet users for selection forcommunication.

In an optional embodiment, the one or more service nodes is configuredto cause the association from the telephone number to the internet userbased on at least one identifier for the internet user selected from agroup consisting of a registered VOIP service screen name of theinternet user; a registered Internet URI describing the internet user; auser identity automatically created and assigned by the system; a username including the telephone number, and a user name including aderivative of the telephone number wherein any of the at least oneidentifier from the above group may be registered on an ENUM domainbased on the telephone number.

In an optional embodiment, the association is based only on: thetelephone number of the telephony subscriber device, and the internetuser.

In an optional embodiment, the association further includes status of aclient proxy.

In an optional embodiment, the system is further configured to: receive,from at least one gateway, a query regarding the telephone number of thetelephony subscriber device; and respond to the at least one gatewaywith information regarding internet users associated with the telephonenumber of the telephony subscriber device.

In an optional embodiment, the information includes zero or moreassociations allowing the telephony subscriber device to connect to oneor more internet users.

In an optional embodiment, the service node is further configured tocreate the association in response to a query from the internet userregarding the telephone number.

According to the teachings of the present embodiment there is provided asystem for supporting communications between an internet user on aninternet-addressed-communications-device and a telephony subscriberdevice, the telephony subscriber device having a corresponding telephonenumber, the system including: at least one gateway configured to receivefrom a service node information regarding internet users associated witha telephone number of the telephony subscriber, the information includesone or more associations, from the telephone number to the internetuser, the associations allowing the telephony subscriber device toconnect to one or more internet users.

In an optional embodiment, the at least one gateway is furtherconfigured to perform a query in response to receiving a communicationsrequest from the telephony subscriber device the query to the servicenode regarding a telephone number of the telephony subscriber. In anoptional embodiment, the at least one gateway is further configured toinitiate communications between the telephony subscriber device and oneor more internet users based on the information. In an optionalembodiment, the at least one gateway is further configured to provideconnectivity between the telephony subscriber device and one or moreinternet users.

In an optional embodiment, the at least one gateway configured to:receive a communications request from the telephony subscriber device,the communications request including the telephone number of thetelephony subscriber device; and cause, by using the received telephonenumber, retrieval of addressing information for the internet user thatis associated with the telephone number, wherein the system isconfigured to use the retrieved addressing information to causeestablishment of a communications connection between the telephonysubscriber device and a device then being used by the internet user,thereby supporting communications between the telephony subscriberdevice and the internet user.

In an optional embodiment, the addressing information for the internetuser includes, but is not limited to: an Internet service user address,an Internet service user name, an Internet URI, an IP address, an IPcommunications endpoint address, and a MAC address.

In an optional embodiment, in response to receiving a communicationsrequest from the telephony subscriber device, the communications requestincluding the telephone number of the telephony subscriber device, thesystem is further configured to: instantiate a client proxy for thetelephone number of the telephony subscriber device, the client proxyoperative to facilitate communications between the telephony subscriberdevice and internet user.

In an optional embodiment, at least one of the gateways is furtherconfigured to: receive information regarding a plurality of internetusers for which addressing information is associated with the telephonenumber, and configure, based on the information received, the clientproxy to facilitate communications between the telephony subscriberdevice and at least one of the internet users for which addressinginformation has been received.

In an optional embodiment, the client proxy includes status of thetelephony subscriber device. In an optional embodiment, the client proxyis configured to publish information for the telephony subscriber deviceand the associated internet users. In an optional embodiment, the clientproxy is configured to translate between communication protocols usedrespectively by the telephony subscriber device and one or more devicesused by the internet user.

In an optional embodiment, if addressing information of a singleavailable internet user is associated with the telephone number, atleast one gateway is configured to: notify the telephony subscriberdevice of a communications possibility; cause establishment of aconnection from the telephony subscriber device to the internet user,and cause the identification of the telephony subscriber device, as acalling party, to the internet user.

In an optional embodiment, if a plurality of available internet users isassociated with the telephone number, at least one gateway is configuredto present to the telephony subscriber device a list of the plurality ofavailable internet users for selection for communication.

According to the teachings of the present embodiment there is provided asystem for supporting communications between a internet user on aninternet-addressed-communications-device and a telephony subscriberdevice, the telephony subscriber device having a corresponding telephonenumber, the system including: at least one gateway configured so thatupon receiving a communications request from the internet user forconnection to the telephony subscriber device the gateway requests anassociation be created from a telephone number of the telephonysubscriber to the internet user, the association allowing the telephonysubscriber device to connect to the internet user.

In an optional embodiment, the system further includes a service nodeconfigured to create associations, wherein the gateway requests theservice node create the association.

According to the teachings of the present embodiment there is provided amethod for supporting communications between an internet user on aninternet-addressed-communications-device and a telephony subscriberdevice, the telephony subscriber device having a corresponding telephonenumber, the method including the steps of: receiving at one or moreservice nodes, the telephone number of the telephony subscriber device;and creating an association from the telephone number to the internetuser, the association allowing the telephony subscriber device toconnect to the internet user.

In an optional embodiment, the telephone number is received in responseto an action by the internet user. In an optional embodiment, theassociation from the telephone number of the telephony subscriber to theinternet user includes the Internet address of the internet user. In anoptional embodiment, the telephone number is provided to the servicenode from a gateway.

In an optional embodiment, the current method includes the steps ofreceiving at a gateway a communications request from the telephonysubscriber device, the communications request including the telephonenumber of the telephony subscriber device; and causing, by using thereceived telephone number, retrieval of addressing information for theinternet user that is associated with the telephone number, wherein thesystem is configured to use the retrieved addressing information tocause establishment of a communications connection between the telephonysubscriber device and a device then being used by the internet user,thereby supporting communications between the telephony subscriberdevice and the internet user.

In an optional embodiment, the addressing information for the internetuser includes but is not limited to: an Internet service user address,an Internet service user name, an Internet URI, an IP address, an IPcommunications endpoint address, and a MAC address.

In an optional embodiment, in response to receiving the communicationsrequest from the telephony subscriber device: instantiating a clientproxy for the telephone number of the telephony subscriber device, theclient proxy operative to facilitate communications between thetelephony subscriber device and internet user.

In an optional embodiment, the current method includes the steps of:receiving, at least one of the gateways, information regarding aplurality of internet users for which addressing information isassociated with the telephone number; and configuring, based on theinformation received, the client proxy to facilitate communicationsbetween the telephony subscriber device and at least one of the internetusers for which addressing information has been received.

In an optional embodiment, the client proxy includes status of thetelephony subscriber device. In an optional embodiment, the client proxyis configured for publishing information for the telephony subscriberdevice and the associated internet users. In an optional embodiment, theclient proxy is configured for translating between communicationprotocols used respectively by the telephony subscriber device and oneor more devices used by the internet user.

In an optional embodiment, the one or more service nodes is furtherconfigured for: causing associations between the telephone number andaddressing information, the addressing information respectivelyassociated with a plurality of internet users; and permitting thetelephony subscriber device to establish communications with at leastone of the plurality of internet users.

In an optional embodiment, if addressing information of a singleavailable internet user is associated with the telephone number, atleast one gateway is configured for: notifying the telephony subscriberdevice of a communications possibility, causing establishment of aconnection from the telephony subscriber device to the internet user;and causing the identification of the telephony subscriber device, as acalling party, to the internet user.

In an optional embodiment, if a plurality of available internet users isassociated with the telephone number, at least one gateway is configuredfor: presenting to the telephony subscriber device a list of theplurality of available internet users for selection for communication.

In an optional embodiment, the one or more service nodes is configuredfor causing the association from the telephone number to the internetuser based on at least one identifier for the internet user selectedfrom a group consisting of: a registered VOIP service screen name of theinternet user; a registered Internet URI describing the internet user, auser identity automatically created and assigned by the system; a username including the telephone number; and a user name including aderivative of the telephone number, wherein any of the at least oneidentifier from the above group may be registered on an ENUM domainbased on the telephone number.

In an optional embodiment, the association is based only on: thetelephone number of the telephony subscriber device, and the internetuser.

In an optional embodiment, the association further includes status of aclient proxy.

In an optional embodiment, the current method includes the steps ofreceiving, from at least one gateway, a query regarding the telephonenumber of the telephony subscriber device; and responding to the atleast one gateway with information regarding internet users associatedwith the telephone number of the telephony subscriber device.

In an optional embodiment, the current method includes the step of:initiating, by the gateway, communications between the telephonysubscriber device and one or more internet users based on one or moreassociations in the information.

In an optional embodiment, the current method includes the step of:facilitating by the gateway, communications between the telephonysubscriber device and one or more internet users.

In an optional embodiment, the information includes zero or moreassociations allowing the telephony subscriber device to connect to oneor more internet users.

In an optional embodiment, the service node is further configured tocreate the association in response to a query from the internet userregarding the telephone number.

According to the teachings of the present embodiment there is provided amethod for supporting communications between an internet user on aninternet-addressed-communications-device and a telephony subscriberdevice, the telephony subscriber device having a corresponding telephonenumber, the method including the steps of sending, from a service node,information regarding internet users associated with a telephone numberof the telephony subscriber; and receiving, to at least one gateway, theinformation, wherein the information includes one or more associations,from the telephone number to the internet user, the associationsallowing the telephony subscriber device to connect to one or moreinternet users.

In an optional embodiment, the at least one gateway is furtherconfigured for performing a query in response to receiving acommunications request from the telephony subscriber device the query tothe service node regarding a telephone number of the telephonysubscriber. In an optional embodiment, the at least one gateway isfurther configured for initiating communications between the telephonysubscriber device and one or more internet users based on theinformation. In an optional embodiment, the at least one gateway isfurther configured for providing connectivity between the telephonysubscriber device and one or more internet users.

In an optional embodiment, the at least one gateway is furtherconfigured for: receiving a communications request from the telephonysubscriber device, the communications request including the telephonenumber of the telephony subscriber device; and causing, by using thereceived telephone number, retrieval of addressing information for theinternet user that is associated with the telephone number, wherein thesystem is configured to use the retrieved addressing information tocause establishment of a communications connection between the telephonysubscriber device and a device then being used by the internet user,thereby supporting communications between the telephony subscriberdevice and the internet user.

In an optional embodiment, the addressing information for the internetuser includes, but is not limited to: an Internet service user address,an Internet service user name, an Internet URI, an IP address, an IPcommunications endpoint address, and a MAC address.

In an optional embodiment, in response to receiving a communicationsrequest from the telephony subscriber device: instantiating a clientproxy for the telephone number of the telephony subscriber device, theclient proxy operative to facilitate communications between thetelephony subscriber device and the internet user.

In an optional embodiment, the current method further includes the stepsof: receiving, at least one of the gateways, information regarding aplurality of internet users for which addressing information isassociated with the telephone number; and configuring, based on theinformation received, the client proxy to facilitate communicationsbetween the telephony subscriber device and at least one of the internetusers for which addressing information has been received.

In an optional embodiment, the client proxy includes status of thetelephony subscriber device. In an optional embodiment, the client proxyis configured for publishing information for the telephony subscriberdevice and the associated internet users. In an optional embodiment, theclient proxy is configured to translate between communication protocolsused respectively by the telephony subscriber device and one or moredevices used by the internet user.

In an optional embodiment, if addressing information of a singleavailable internet user is associated with the telephone number, atleast one gateway is configured for notifying the telephony subscriberdevice of a communications possibility; causing establishment of aconnection from the telephony subscriber device to the internet user,and causing the identification of the telephony subscriber device, as acalling party, to the internet user.

In an optional embodiment, if a plurality of available internet users isassociated with the telephone number, at least one gateway is configuredfor presenting to the telephony subscriber device a list of theplurality of available internet users for selection for communication.

In the system and method of the present embodiment, information caninclude parameters selected from the group consisting of: net contacts;and an Internet address of the internet users. In the system and methodof the present embodiment, information can include two or moreassociations, and a selection process is used to resolve which one ormore internet users to initiate connections for the telephony subscriberdevice. In the system and method of the present embodiment whereininformation includes two or more associations, a selection process isused to resolve which one or more internet users to initiate connectionsfor the telephony subscriber device, wherein the selection processincludes using secondary caller line identity.

In the system and method of the present embodiment, the association fromthe telephone number of the telephony subscriber to the internct usercan include an Internet address of the internet user.

In the system and method of the present embodiment, the telephone numberof the telephony subscriber can be selected from a group consisting ofdecimal number according to ITU recommendation E.164; and a calling lineidentifier (CLI) of the telephony subscriber device.

In the system and method of the present embodiment, the association canbe a unidirectional association from the telephone number of thetelephony subscriber to the internet user.

In the system and method of the present embodiment, the telephone numbercan be a unique telephone number identifying the telephony subscriberdevice, and the association between the telephone number and the user isdependent on the telephone number without necessarily being dependent ona gateway telephone number.

According to the teachings of the present embodiment there is provided asystem for interconnecting telephony user domains including: a serviceaccess gateway between a first numerically addressed telephony usernetwork and a second alphanumerically addressed telephony user network,said service access gateway configured to automatically associate afirst user calling a gateway from the first network, with other users onthe second network who have previously identified the first user on thefirst network, thereby reverse mapping a numerical calling line identitypresented at the gateway by the first numerically addressed telephonyuser network into the associated alphanumerical user identities on thesecond alphanumerically addressed telephony user network; and presentand selectively bridge connections between the first user on the firstnumerically addressed telephony user network and associated users on thesecond alphanumerically addressed telephony user network.

According to the teachings of the present embodiment there is provided amethod for facilitating communications between a respective first andsecond telephony domain including the steps of receiving, at a servicenode associated with the first domain, from a user identity registeredin the first domain, a unique telephone number identifying a second userin the second domain; associating addressing information of the user inthe first domain, with the received unique telephone number; receiving aconnection request from the second domain, originating from a telephonydevice identified by the received unique telephone number; routing theconnection request to a communications gateway between the first andsecond telephony domains; retrieving addressing information to locatethe user in the first domain who is associated with the calling lineidentity of the second user in the second domain; and using theretrieved addressing information, facilitating communication from thesecond user in the second domain to the user in the first domain.

According to the teachings of the present embodiment there is provided acomputer-readable storage medium having embedded thereoncomputer-readable code for supporting communications between an internetuser on an internet-addressed-communications-device and a telephonysubscriber device, the telephony subscriber device having acorresponding telephone number by: receiving at one or more servicenodes, the telephone number of the telephony subscriber device; andcreating an association from the telephone number to the internet user,the association allowing the telephony subscriber device to connect tothe internet user.

In an optional embodiment, the computer-readable storage medium hasembedded thereon additional computer-readable code for: receiving at agateway a communications request from the telephony subscriber device,the communications request including the telephone number of thetelephony subscriber device; and causing, by using the receivedtelephone number, retrieval of addressing information for the internetuser that is associated with the telephone number, wherein the system isconfigured to use the retrieved addressing information to causeestablishment of a communications connection between the telephonysubscriber device and a device then being used by the internet user,thereby supporting communications between the telephony subscriberdevice and the internet user.

According to the teachings of the present embodiment there is provided acomputer-readable storage medium having embedded thereoncomputer-readable code for supporting communications between an internetuser on an internet-addressed-communications-device and a telephonysubscriber device, the telephony subscriber device having acorresponding telephone number by: receiving a communications requestfrom the telephony subscriber device, the communications requestincluding the telephone number of the telephony subscriber device; andcausing, by using the received telephone number, retrieval of addressinginformation for the internet user that is associated with the telephonenumber, wherein the retrieved addressing information is used to causeestablishment of a communications connection between the telephonysubscriber device and a device then being used by the internet user,thereby supporting communications between the telephony subscriberdevice and the internet user.

According to the teachings of the present embodiment there is provided acomputer program that can be loaded onto a server connected through anetwork to a client computer, so that the server running the computerprogram constitutes a service node in a system according to any one ofthe system embodiments. According to the teachings of the presentembodiment there is provided a computer program that can be loaded ontoa computer connected through a network to a server, so that the computerrunning the computer program constitutes a service node in a systemaccording to any one of the system embodiments. According to theteachings of the present embodiment there is provided a computer programthat can be loaded onto a server connected through a network to a clientcomputer, so that the server running the computer program constitutes agateway in a system according to any one of the system embodiments.According to the teachings of the present embodiment there is provided acomputer program that can be loaded onto a computer connected through anetwork to a server, so that the computer running the computer programconstitutes a gateway in a system according to any one of the systemembodiments.

According to the teachings of the present embodiment there is provided asystem for supporting communications between a first user and at leastone other user including: a service node configured to: receive from thefirst user a handle; create, based on the handle, a first registrationto the first user, receive from at least one other user the handle; andcreate, based on the handle, at least one additional registrationrespectively to the at least one other user, wherein when the firstregistration and at least one additional registration are active, aconnection is invited between the first user and the at least one otheruser, the connection facilitating a direct communication between thefirst user and the at least one other user.

In an optional embodiment, after terminating the connection, the firstregistration and the at least one additional registration remain active,thereby facilitating re-connection of Internet communication devicesregistered with the handle. In an optional embodiment, registrations areuniversal resource locators (URLs). In an optional embodiment, a domainis created linking the URL to the first user. In an optional embodiment,registration includes registering, based on the handle, a VoIPcommunications endpoint at the network address of the user.

According to the teachings of the present embodiment there is provided asystem for supporting communications between aa first user and at leastone other user including: a service node configured to: receive from thefirst user a handle; register, based on the handle, a first IPcommunications endpoint at the network address of the first user,receive from at least one other user the handle; and register, based onthe handle, at least a second IP communications endpoint at therespective network address of the at least one other user; wherein thefirst and at least a second IP communications endpoints are configuredto permit inviting a connection between endpoints registered with thehandle.

In an optional embodiment, after terminating the connection, the firstIP communications endpoint and at least a second IP communicationsendpoints remain active, thereby facilitating re-connection of Internetcommunication devices registered with the handle. In an optionalembodiment, the IP communications endpoints are Universal ResourceLocators (URLs). In an optional embodiment, the Universal ResourceLocators (URLs) are the same.

According to the teachings of the present embodiment there is provided amethod for supporting communications between a first user and at leastone other user, the method including the steps of receiving, at aservice node, from the first user a handle; creating, based on thehandle, a first registration to the first user; receiving, at theservice node, from at least one other user the handle; and creating,based on the handle, at least one additional registration respectivelyto the at least one other user wherein when the first registration andat least one additional registration are active, a connection is invitedbetween the first user and the at least one other user, the connectionfacilitating a direct communication between the first user and the atleast one other user.

In an optional embodiment, after terminating the connection: maintainingthe first registration and the at least one additional registration inan active state, thereby facilitating re-connection of Internetcommunication devices registered with the handle. In an optionalembodiment, registration includes: registering, based on the handle, aVoIP communications endpoint at the network address of the user.

According to the teachings of the present embodiment there is provided amethod for supporting communications between a first user and at leastone other user, the method including the steps of: receiving, at aservice node, from the first user a handle; registering, based on thehandle, a first VoIP communications endpoint at the network address ofthe first user; receiving from at least one other user the handle; andregistering, based on the handle, at least a second VoIP communicationsendpoint at the respective network address of the at least one otheruser, wherein the first and at least a second VoIP communicationsendpoints are configured to permit inviting a connection betweenendpoints registered with the handle.

In an optional embodiment, after terminating the connection: maintainingthe first VoIP communications endpoint and at least a second VoIPcommunications endpoints in an active state, thereby facilitatingre-connection of Internet communication devices registered with thehandle.

According to the teachings of the present embodiment there is provided acomputer-readable storage medium having embedded thereoncomputer-readable code for supporting communications between a firstuser and at least one other user by: receiving, at a service node, fromthe first user a handle; creating, based on the handle, a firstregistration to the first user, receiving, at the service node, from atleast one other user the handle; and creating, based on the handle, atleast one additional registration respectively to the at least one otheruser; wherein when the first registration and at least one additionalregistration are active, a connection is invited between the first userand the at least one other user, the connection facilitating a directcommunication between the first user and the at least one other user.

According to the teachings of the present embodiment there is provided acomputer-readable storage medium having embedded thereoncomputer-readable code for supporting communications between a firstuser and at least one other user by: receiving, at a service node, fromthe first user a handle; registering, based on the handle, a first VoIPcommunications endpoint at the network address of the first user;receiving from at least one other user the handle; and registering,based on the handle, at least a second VoIP communications endpoint atthe respective network address of the at least one other user, whereinthe first and at least a second VoIP communications endpoints areconfigured to permit inviting a connection between endpoints registeredwith the handle.

According to the teachings of the present embodiment there is provided acomputer program that can be loaded onto a server connected through anetwork to a client computer, so that the server running the computerprogram constitutes a service node in a system according to any one ofthe system embodiments. According to the teachings of the presentembodiment there is provided a computer program that can be loaded ontoa computer connected through a network to a server, so that the computerrunning the computer program constitutes a service node in a systemaccording to any one of the system embodiments.

According to the teachings of the present embodiment there is provided acontroller for supporting communications between an internet user on aninternet-addressed-communications network and a telephony subscriber,the controller configured for: obtaining a telephone number associatedwith the telephony subscriber; determining whether an internet useridentity corresponding to the telephone number identity is registeredwith the internet-addressed-communications network; if no internet useridentity corresponding to the telephone number identity is registered,automatically creating and registering with theinternet-addressed-communications network, a new internet user identitycorresponding to and using the telephone number identity and

recording the internet user as a contact for the new internet useridentity corresponding to the telephone number identity.

According to the teachings of the present embodiment there is provided amethod for supporting communications between an internet user on aninternet-addressed-communications network and a telephony subscriber,the method including: obtaining a telephone number associated with thetelephony subscriber, determining whether an internet user identitycorresponding to the telephone number identity is registered with theinternet-addressed-communications network; if no internet user identitycorresponding to the telephone number identity is registered,automatically creating and registering with theinternet-addressed-communications network, a new internet user identitycorresponding to and using the telephone number identity and recordinguser as a contact for the new internet user identity corresponding tothe telephone number identity.

In an optional embodiment, obtaining the telephone number includes theinternet user entering the telephone number. In an optional embodiment,the internet user enters the telephone number as a contact screen name.

In an optional embodiment, if an internet user identity corresponding tothe telephone number identity is registered with theinternet-addressed-communications network, recording the internet useras a contact for the internet user identity corresponding to thetelephone number identity.

In an optional embodiment, wherein the new internet user identity isautomatically created and registered with a screen name including thetelephone number identity.

In an optional embodiment, obtaining the telephone number includes:retrieving a plurality of recorded contacts for the internet user, anddetermining any phone numbers in any retrieved contacts for the internetuser.

In an optional embodiment, the current method further includes using thenew internet user identity to establish communications between thetelephony subscriber and the internet user, thereby supportingcommunications between the telephony subscriber and the internet user.

In an optional embodiment, the current method further includes the stepof configuring a client proxy to support communications between at leastone telephony subscriber, having a telephone number, and the internetuser.

In an optional embodiment, the client proxy translates communicationprotocols used respectively by the telephony subscriber device and oneor more internet-addressed-communications-devices used by the internetuser.

In an optional embodiment, the addressing information of an availableinternet user is associated with the telephone number, the methodfurther includes: notifying the telephony subscriber of a communicationspossibility; causing establishment of a connection to the internet user;and causing the presenting of the identity of the telephony subscriberas a calling party.

In an optional embodiment, presenting to the telephony subscriber a listof internet users on the internet-addressed-communications network, fromwhich list the telephony subscriber can select an internet user withwhom to establish communications.

According to the teachings of the present embodiment there is provided afor supporting communications between an internet user on aninternet-addressed-communications network and a telephony subscriber,the controller configured for: allowing the internet user to request acommunications connection with a telephone user by dialing a phonenumber of the telephone user, determining whether the telephonysubscriber is in communication with the internet-addressedcommunications network by: assembling a client service user name usingthe dialed phone number; and querying a service registry with theassembled client user name for an address of an internet client proxyfor the telephony subscriber; if the service registry does not containan active address for a proxy of the telephony subscriber, then routingthe communication connection request to an Internet hosted InteractiveVoice Response (IVR) messaging system, allowing the user to use the IVRmessaging system to create a message for the telephone subscriber and ifthe service registry does contain an active internet address for aclient proxy of the telephony subscriber, routing the communicationconnection request from the internet user to an internet addressassociated with the client proxy for telephony subscriber.

According to the teachings of the present embodiment there is provided amethod for supporting communications between an internet user on aninternet-addressed-communications network and a telephony subscriber,the method including: allowing the internet user to request acommunications connection with the telephony subscriber by dialing aphone number of the telephone user; determining whether the telephonysubscriber is in communication with the internet-addressedcommunications network by: assembling a client service user name usingthe dialed phone number, and querying a service registry with theassembled client user name for an address of an internet client proxyfor the telephony subscriber; if the service registry does not containan active address for a proxy of the telephony subscriber, then routingthe communication connection request to an Internet hosted InteractiveVoice Response (IVR) messaging system, allowing the internet user to usethe IVR messaging system to create a message for the telephonesubscriber; and if the service registry does contain an active internetaddress for a client proxy of the telephony subscriber, routing thecommunication connection request from the internet user to an IP addressassociated with the client proxy for telephony subscriber.

In an optional embodiment, after routing the communication connectionrequest from the internet user to an IP address associated with thetelephony subscriber, notifying the telephony subscriber of an incomingcommunication connection request from the internet user. In an optionalembodiment, the IP address associated with the telephony subscriber isan IP address of the Internet client proxy for the telephony subscriber.

In an optional embodiment, the current method further includes the stepof awaiting acceptance of the incoming communication connection request,from the telephony subscriber.

In an optional embodiment, the telephony subscriber selects apredetermined key to accept the incoming communication connectionrequest. In an optional embodiment, the telephony subscriber issues apredetermined voice command to accept the incoming communicationconnection request. In an optional embodiment, the telephony subscriberdeclines to accept the incoming communication connection request, by notresponding to the announcement. In an optional embodiment, if thetelephony subscriber does not accept the incoming connection request,the connection is rerouted to an internet messaging platform.

In another optional embodiment, the current method includes if telephonysubscriber accepts the incoming connection request from user, placingany currently active connection on hold and connecting the user.

According to the teachings of the present embodiment there is provided acomputer-readable storage medium having embedded thereoncomputer-readable code for supporting communications between an internetuser on an internet-addressed-communications network and a telephonysubscriber by: obtaining a telephone number associated with thetelephony subscriber; determining whether an internet user identitycorresponding to the telephone number identity is registered with theinternet-addressed-communications network; if no internet user identitycorresponding to the telephone number identity is registered,automatically creating and registering with theinternet-addressed-communications network, a new internet user identitycorresponding to and using the telephone number identity; and recordingthe internet user as a contact for the new internet user identitycorresponding to the telephone number identity.

According to the teachings of the present embodiment there is provided acomputer program that can be loaded onto a server connected through anetwork to a client computer, so that the server running the computerprogram constitutes a controller in a system according to any one of thesystem embodiments. According to the teachings of the present embodimentthere is provided a computer program that can be loaded onto a computerconnected through a network to a server, so that the computer runningthe computer program constitutes a controller in a system according toany one of the system embodiments.

According to the teachings of the present embodiment there is provided acomputer-readable storage medium having embedded thereoncomputer-readable code for supporting communications between an internetuser on an internet-addressed-communications network and a telephonysubscriber by: allowing the internet user to request a communicationsconnection with the telephony subscriber by dialing a phone number ofthe telephone user, determining whether the telephony subscriber is incommunication with the IP-addressed communications network by:assembling a client service user name using the dialed phone number; andquerying a service registry with the assembled client user name for anaddress of an internet client proxy for the telephony subscriber; if theservice registry does not contain an active address for a proxy of thetelephony subscriber B, then routing the communication connectionrequest to an internet hosted Interactive Voice Response (IVR) messagingsystem, allowing the internet user to use the IVR messaging system tocreate a message for the telephone subscriber, and if the serviceregistry does contain an active internet address for a client proxy ofthe telephony subscriber, routing the communication connection requestfrom the internet user to an internet address associated with the clientproxy for telephony subscriber.

According to the teachings of the present embodiment there is provided acomputer program that can be loaded onto a server connected through anetwork to a client computer, so that the server running the computerprogram constitutes a controller in a system according to any one of thesystem embodiments. According to the teachings of the present embodimentthere is provided a computer program that can be loaded onto a computerconnected through a network to a server, so that the computer runningthe computer program constitutes a controller in a system according toany one of the system embodiments.

According to the teachings of the present embodiment there is provided acontroller for authenticating a user on an Internet device using atelephony subscriber device of the user, the telephony subscriber devicehaving a corresponding telephone number, the controller configured forreceiving, at a gateway, the telephone number via an Internet networkfrom the user, initiating a phone call from the gateway via a telephonynetwork to the user's telephony subscriber device, the phone callincluding a call-back telephone number; initializing a countdown timerwith a given time; and authenticating the user to the Internet device ifa phone call is received at the gateway from the user's telephonysubscriber device; and the phone call is received before the countdowntimer expires.

According to die teachings of the present embodiment there is provided amethod of authenticating a user on an Internet device using a telephonysubscriber device of the user, the telephony subscriber device having acorresponding telephone number, the method including the steps ofreceiving, at a gateway, the telephone number via an Internet networkfrom the user; initiating a phone call from the gateway via a telephonynetwork to the user's telephony subscriber device, the phone callincluding a call-back telephone number, initializing a countdown timerwith a given time; and authenticating the user to the Internet device ifa phone call is received at the gateway from the user's telephonysubscriber device; and the phone call is received before the countdowntimer expires.

In an optional embodiment, the telephony network is a POTS (plain oldtelephone system) network. In another optional embodiment, the phonecall from the gateway via a telephony network to the user's telephonysubscriber device is a missed call. In another optional embodiment, thecallback telephone number is randomly generated on the gateway prefix.In another optional embodiment, the countdown timer automatically countsdown to zero, thereby timing the phone call received at the gateway fromthe user's telephony subscriber device. In another optional embodiment,if the countdown timer expires bebfore phone call is received at thegateway from the user's telephony subscriber device, authenticationfails for the user. In another optional embodiment, if the phone callreceived at the gateway from the user's telephony subscriber device isto a telephone number other than the callback number, authenticationfails for the user. In another optional embodiment, if the phone callreceived at the gateway to the call-back number is from other than theuser's telephony subscriber device, authentication fails for the user.

According to the teachings of the present embodiment there is provided acomputer-readable storage medium having embedded thereoncomputer-readable code for authenticating an internet user on aninternet device using a telephony subscriber device of the internetuser, the telephony subscriber device having a corresponding telephonenumber, by: receiving, at a gateway, the telephone number via aninternet network from the internet user, initiating a phone call fromthe gateway via a telephony network to the internet user's telephonysubscriber device, the phone call including a call-back telephonenumber; initializing a countdown timer with a given time; andauthenticating the internet user to the internet device if a phone callis received at the gateway from the internet user's telephony subscriberdevice; and the phone call is received before the countdown timerexpires.

According to the teachings of the present embodiment there is provided acomputer program that can be loaded onto a server connected through anetwork to a client computer, so that the server running the computerprogram constitutes a controller in a system according to any one of thesystem embodiments. According to the teachings of the present embodimentthere is provided a computer program that can be loaded onto a computerconnected through a network to a server, so that the computer runningthe computer program constitutes a controller in a system according toany one of the system embodiments.

BRIEF DESCRIPTION OF DRAWINGS

For a better understanding of the embodiments described in thisapplication, reference should be made to the Detailed Description below,in conjunction with the following drawings in which like referencenumerals refer to corresponding parts throughout the figures.

FIG. 1: Abstracted core service elements in a contact embodiment.

FIG. 2: Detailed logical interaction between core elements.

FIG. 3: Tele IP Presence Protocol fundamentals.

FIG. 4: Tele IP Presence Protocol aliased.

FIG. 5: Sample Stored Procedures transforming contacts.

FIG. 6: VOIP originated call flow to phones OFF net.

FIG. 7: VOIP originated call flow to phones ON net.

FIG. 8: VOIP virtual client proxy instantiation.

FIG. 9: VOIP ON/OFF net switching flow chart.

FIG. 10: Delivering OFF net call notifications.

FIG. 11: Abstracted core service elements in a dial stream embodiment.

FIG. 12: Basic VOIP originated dial stream stepladder.

FIG. 13: Basic VOIP redirected call flow chart.

FIG. 14: Basic VOIP terminated switching stepladder.

FIG. 15: Timed VOIP terminated switching.

FIG. 16: Sample Menu Play list flow chart.

FIG. 17: Virtual Realm service abstraction.

FIG. 18: Universal Edge caching data model.

FIG. 19: Paired OFF net ring back matrix.

FIG. 20: Paired OFF net ring back stepladder.

FIG. 21: Abstracted core service elements in a symbolic prefixembodiment.

FIG. 22: Detailed logical interaction between prefixed core elements.

FIG. 23: Prefixed VOIP terminated call flow chart.

FIG. 24: Prefixed VOIP terminated switching stepladder.

FIG. 25: E164 dial plan virtualization schema.

FIG. 26: Randomized Telephone number authentication flow chart.

FIG. 27: Randomized Telephone number entity relationships.

FIG. 28: Returned Telephone number authentication flow chart.

FIG. 29: Returned Telephone number authentication entity relationships.

FIG. 30: Global IP phone number assignment in one embodiment.

FIG. 31: Example Telephone pairing over IP.

FIG. 32: Example Web social telephony mash up.

FIG. 33: Example Instant callback service channel.

FIG. 34: Diagram highlighting core net versus cellular culture.

FIG. 35: Behavioral economics converging technology and culture.

FIG. 36: Newly disclosed “on and offnet” connection flows.

FIG. 37: Price elasticity and the penny gap.

FIG. 38: Graph plotting social parabolic geometry.

FIG. 39: FRISB World Wide Web service embodiment.

FIG. 40: FRISB universal color coded signaling state management.

FIG. 41: FRISB disc edge caching over dynamic DNS.

FIG. 42: FRISB disc dynamic DNS zone delegation.

FIG. 43: FRISB sample zone A records.

FIG. 44: NANO (Number As Named Object) “1:many:1” switching.

FIG. 45: ALTO (Authenticated Line To Owner) “many:1” switching.

FIG. 46: UNNO (Universal Net Number) “1:1” switching.

FIG. 47: PICO (Paired Index Coupling) “many.many” switching.

FIG. 48: SOLO (Switched On Line Origination) “1:many” switching.

DETAILED DESCRIPTION

Programmatically reversing numerical line identity presented at acommunications services gateway into named IP Telephony users with“prior association”, delivers dynamic “reverse address resolution”switching connections from ground to cloud, permitting any conventionaltelephone to dial and connect to any associated IP Telephony endpoint inthe world, without changes to the conventional telephone. Reversing lineidentity into associated named users bridges both the addressability andeconomic divide between mass conventional “paying” (mobile and fixed)and “free” IP Telephony networks.

A system for supporting communications between a user on anIP-addressed-communications-device and a telephony subscriber device,the telephony subscriber device having a corresponding telephone number,includes: one or more service nodes configured to: receive from the userthe telephone number of the telephony subscriber device and create anassociation from the telephone number to the user, wherein theassociation allows the telephony subscriber device to connect to theuser.

TERMS

In the context and scope of the embodiments the term “Voice OverInternet Protocol” (VOIP) is generally used synonymously andinterchangeably with “Internet”, “IP Telephony”, “Endpoint”, “UserAgent”, “Session Initiation Protocol (SIP)” and other Internet TelephonyProtocols Methods and Systems.

The public switched telephone network (PSTN) is the network of theworld's public circuit-switched telephone networks, inter-connected byswitching centers. In the context of this document, the term telephonycommunications device generally refers to an end-user device on thePSTN. End user devices include, but are not limited to land (fixed)-linetelephones, mobile telephones, fax machines, and modems.

The terms “Telephone”, “Phone”, “Mobile”, “Cell”, “telephony subscriberdevice” and other communication devices (portable or otherwise) aregenerally used synonymously and interchangeably with, for example, PLMN(Public Land Mobile Networks) and PSTN (Public Switched TelephonyNetworks) compatible devices. A telephony subscriber device operates viaa network that is separate from the network on which an internet deviceoperates. A variety of separate telephony networks can be used.Typically, the telephony network is a POTS (plain old telephone service)network, having connections between telephony subscriber devices(telephones) and gateways that are physically separate from theconnections between the Internet device and the gateways. As theprotocols are different between the Internet network and telephonynetwork.

In the context of this document, the term “internet” generally refers toconnection of a network to other networks. The resulting system ofinterconnected networks is called an internetwork, or simply aninternet. The most notable example of an internet is the Internet, aworldwide network of networks based on many underlying hardwaretechnologies, but unified by an internetworking protocol standard, theInternet Protocol Suite, often also referred to as TCP/IP. The terms“internet” and “Internet” may be used interchangeably in this document.

The term “alphanumeric” may be used interchangeably with “alpha”, thatis, an identity comprising both alphabetical characters and numericdigits. While alphanumeric terms typically include at least alphabeticalcharacters mixed with numbers and/or symbols, alphanumeric includesreference to terms that can be all alphabetical characters, all numericcharacters, symbols, and a mixture of character types. The term“numeric”, refers to conventional telephone numbers having numericaldigits that may be symbolically prefixed and/or mixed with standardtelephone keypad symbols, for example, the Star (*) or Pound (#) keysand combinations thereof.

In the context of this document, the term “internet address” generallyrefers to the address and/or designator to identify where a device islocated on a network. One skilled in the art will be aware of thecontext and use of alternative terms, including but not limited to“network address”, “physical address”, “hardware address”, “address ofthe PC”, and “address of the internet device”.

The term “Gateway” described herein could be any gateway, including forexample, a “PSTN Internet Gateway”, “Trunking Gateway”, “Media Server”,“Proxy Server”, IMS (Internet Multimedia Subsystem), IPBX (InternetProtocol Branch Exchange), “Soft Switch”, Signaling Controllers andother such telecommunication interconnecting service nodes.

The term “Registry” could be used synonymously and interchangeably with“database registry”, “service registry”, “Registrar”, B2BUA (Back toBack User Agent), Inbound Proxy, VOIP service platforms such asAsterisk™, FreeSWITCH™ and other such communication platforms. Registryand Gateway may logically be one and the same platform; they may bephysically separate nodes and combinations thereof.

The term PC (Personal Computer) may be used synonymously andinterchangeably with any Internet Connected Device, such as, but notlimited to, computing devices or any device (portable, handheld orotherwise) capable of connecting to a network (e.g., the Internet).

In the context of this document, the term “internet communicationsdevice” generally refers to devices on an internet capable of providingtelephony services, typically but not limited to voice communications.One skilled in the art will realize that the terms can be usedalternatively, depending on context: Internet communications devices,VoIP phone, computer, subscriber of a VoIP network, and generallyIP-addressed-communications-devices. In this document, internet devicesmay be located on a network, an internet, or the Internet, as will beobvious to one skilled in the art.

The term “Client” described herein could be used synonymously andinterchangeably with VOIP client, “User Agent” (UA), SIP phone,“Softphone”, “Webphone”, Adobe Flash™ Phones, SUN Java™ Phones, Skype™clients and other VOIP capable technologies and telephones that may bedownloadable phones implemented in software code, phones embedded inmultifunctional devices and standalone IP telephone devices. The term“Client” could also generally include any client in a client/serverarrangement.

The term “ON net” may describe a telephone that is connected to aGateway between Telephone and Internet networks. For example, when atelephone is “ON net” there is an associated VOIP proxy representing thetelephone as a “numbered VOIP client”, where the client name comprisesthe phone number, and where a proxy IP Address and Port are assigned andregistered to reference and locate the active client endpoint.

The term “OFF net” may describe a telephone (or communication device)that is disconnected from a Gateway, and as such there may be no proxyassociated with the telephone number. Consequently the telephone, forexample, cannot be located “ON net”, as there is no recorded active IPendpoint in a VOIP service registry that can be referenced.

The capital character “A” used throughout this disclosure, may describeinternet users in the singular and the plural, and the capital character“B”, may describe the telephone with corresponding telephone number “B#”with which User A wishes to establish contact. In many instances, A maytherefore refer to the connection “source” (the caller) and B may referto the “destination” (the called or call recipient). User (A) having aninternet-addressed-communications-device is typically inside a VoIPnetwork, in contrast to telephony subscriber device (B) that istypically inside a conventional PSTN. In other words, B is outside, or anon-subscriber of, the VoIP network of A.

However, since select methods and systems disclosed describe adiscontinuity between completing the connection between A and B, in thatin selected embodiments telephone user B reverse establishescommunication via a services gateway C back to the associated internetuser A, in such cases technically B may be the “source” and A may be the“destination”.

In communication networks, an originating endpoint may be designated as“party A” and a terminating endpoint may be designated as “party B”,however in order to maintain logical consistency in describing thesystems and methods disclosed, while B is still the intended connectiondestination and A the source, the physical connection establishment mayactually reverse the relationship and the endpoints.

This is because selected embodiments logically require party A todisclose “telephone number identifying information” to the serviceprovider permitting party B to originate and complete an inboundconnection back to A. In one embodiment this identifying information isa numerical identity of a communications device utilized on network B.In another embodiment this identification information is the numericalidentity of a communications device logically coupled to the servicesprovided on network A (for example, the authenticated telephone numberassociated with internet user A).

Legends applied to the accompanying figures may include:

-   -   “The cloud”, representing the IP network.    -   “The telephone”, representing legacy Fixed and Mobile telephony        networks.    -   “The pots icon”, plain old telephone system, representing a        telephony capable device, without any special software        capability required.

The enumerated circles (1 2 3 4 5 6 7 8 9 10) mark logical servicesteps. In describing steps, Figure X is abbreviated “FX”. The “A B C D”nomenclature and associated symbols (@ # & *) denote core service nodesand functionality, defined herein as:

A: an alphanumeric “named” VOIP user identity (example:ak@starlogik.com) on an alphanumeric telephony domain;

B: a telephony subscriber device on a numeric telephony domain;

C: a gateway interconnecting the alphanumeric and numeric telephonydomains; and

D: a service node and/or registry storing A and B user associations andVOIP registered endpoints.

These nodes may include symbolically:

@ (“at”) an IP address and port of a VOIP user A;

# (“pound”) a conventional telephone number B, (for example:14154125111);

& (“and”) an association and/or database relationship between B and A;and

* (“star”) a telephony switching element within C.

Note that the use of the “A C D” nomenclature and associated symbols (@# & *) in the claims are for clarity, and should not be considered aslimiting the invention to a specific embodiment.

FIG. 1

FIG. 1 (F1) illustrates the system overview which shows abstracted coreservice elements in context with each other.

F1 shows two telephony domains, a first numerically addressedconventional fixed/mobile telephony network (B# plain old telephone) anda second alphanumeric addressed Internet network (A@ PC in cloud) withinterconnecting Gateway (C*). Logical entity relationships and servicesteps include:

F1 Step1. Internet user A@ who is a user on the second alphanumericallyaddressed IP network, enters the telephone number B# of a contact on thefirst numerically addressed network, into VOIP client software. The VOIPclient software may be resident on user A's Internet device or in thenetwork.

The VOIP service records the telephone number B# in association with theInternet user identity A@, thus binding and associating internet user A@with telephone number B# in a database registry on service node D, inparticular creating an association from the telephone number B# to theInternet user identity A@. In the context of this document, telephonenumbers of a numerically addressed telephony subscriber device B aremore generally referred to by the term “identifier the telephonysubscriber device B#”. A telephone number B# is typically a uniquenumber identifying a corresponding telephony subscriber device (B).

The telephone user receives a call on telephone B indicating a systemcall back number (not shown). The system call back number is preferablya general number for all telephone users B in an area, and notspecifically tied to the Internet user A@.

F1 Step2. Telephone user B dials the phone number of service accessgateway C.

F1 Step3. Registry D is queried by gateway C on the caller line identityB# to determine the associated VOIP contact for user A.

F1 Step4. Gateway C resolves user address A@ to the current IP endpointwhere its VOIP client is registered, which could be through login orother known methods, ringing the destination A with caller line identityB#.

Thus a named VOIP user A@ may be bound to telephone number B# with whomcommunication is desired, in a service registry D, and when B dials atelephony access gateway C then connecting the caller B back to the VOIPuser A@. The vector:

AB: BC: BA

Which yields a reverse switching matrix that may be symbolicallyrepresented as:

$\begin{matrix}{A\mspace{14mu} B\mspace{14mu} D} \\{B\mspace{14mu} C\mspace{14mu} D} \\{D\mspace{14mu} B\mspace{14mu} A}\end{matrix}$

Where registry D is the persistent data store binding A and B.

Thus telephony connections are switched on “source”, the calleridentity, rather than switching purely on dialed “destination”, thecalled identity, thereby reverse mapping the conventional E164 dial planof the caller. This reverse switching permits alphanumeric contact fromany legacy telephone without change, thus enabling legacy telephonesinto “future IP compatible devices”. That is, any conventional telephonedesigned before IP telephony was introduced, phones that have a numericonly keypad and phones without any specialized software capability, arepermitted to dial and connect to any IP telephony endpoint withoutmodifying the handset.

This “IP retrofitting” that delivers backward device compatibility, ismade possible by hooking the new service functionality into aconventionally established call to a gateway access node. Any InternetVOIP user who identifies a conventional telephone number as contact,becomes instantly “discoverable” to the same calling telephone identitypresenting at the gateway to the IP cloud.

This reversed switching algorithm can be used to advance IP Telephonyconvergence in that it permits any conventional telephone user to dial,programmatically discover, and connect to any VOIP user in the world whohas previously identified them by phone number.

By entering and associating conventional telephone numbers of contactsthey wish to establish communication with, an Internet andalphanumerically addressed client thus becomes “reverse addressable” onthe conventional numeric keypad.

On identifying telephone contacts, VOIP users can reverse identifythemselves to the same, thereby conclusively resolving the conundrum“how to address alphanumeric connections on a numeric dial pad” (byaddressing the numeric telephone counterpart on the alphanumeric devicein advance).

This “reverse switching singularity”, in logically transforming alllegacy telephones into IP compatible devices without physicallymodifying the telephone, delivers a defining VOIP service virtualizationto “paying customers”, whose online telephone numbered accounts aresystematically created, and whose online contacts are automaticallyhydrated by and with the users who identify them.

Since the data that comprises online telephony contact can persist in aService Registry, the “reversed calling line identity” now defines therelationship between VOIP (named) users and their associated (numeric)telephone contacts, making it possible to reverse connect the latterwith the former on extracting call line identity at a local gatewayinterconnecting the two domains.

FIG. 2

FIG. 2 shows detailed logical interaction between core elements.

F2 Step1. Internet user A(30) enters the telephone number B(10) of acontact into the VOIP client software. In one embodiment the telephonenumber B is entered into the client contact list and thus identifiedprior to actually dialing the number. In another embodiment user A callstelephone B by entering the phone number using a software telephone dialpad or using the numeric keys on the PC keyboard and B is thusidentified on dialing.

In the present embodiment being described and illustrated, the telephonenumber is entered as a conventional VOIP contact with the exceptionbeing that the “screen name” is set as the phone number. That istelephone contact B is recorded by entering their “number as namedobject” (NANO) which the system then transforms into a corresponding“virtual numbered VOIP client”, accessible only to a matching telephoneidentity calling a service gateway.

Describing this NANO client creation in greater detail, by accepting aphone number as the named contact, a “numbered VOIP name space” andcommunity may be canonically captured and generated. And while anyconventional VOIP user is permitted to generate the NANO client, byentering its phone number, “unilaterally assisting them into creationwithout permission”, only a telephone with a matching identity mayaccess it.

While selected embodiments imply the “automatic acceptance” of any VOIPuser A into the contact list of NANO client B, basic service softwarefunctionality may permit telephone user B to selectively accept ordecline VOIP contacts as they are automatically inserted. NANO clientsmay also be alerted to their automatic account creation, advised of thelocal Gateway telephone access number and notified of contact insertion.These alerts may be delivered via any number of bearers, including SMSand email (when the latter address is included by the identifying netcontact). One such welcoming text display could read:

welcome to nano.

from: ak@starlogik.com

call me on 17132421581

Similarly offering NANO account “management functions” will be evidentto the skilled artisan. For example, the ability to “PIN protect”telephone access to the B NANO client. In one such embodiment, thesystem may automatically generate a PIN password that is securelydelivered to the B party on their NANO client creation (when the firstVOIP contact A enters the B phone number as described).

Alternatively an automated calling attendant may dial and welcome theclient reading the PIN to the B telephone owner. Proxy hosted menus andservices presented to the connected B party may use standard voicescripting and rendering protocols such as VoXML, to transport voicelogic from the application to the Gateway, where IVR (Interactive VoiceResponse) then generates and manages the audio interface presented tothe caller. Once such service may allow PIN bypassing (deactivation) forseamless telephony access to the NANO client.

Further, similar basic software functionality in the NANO B client maypermit telephone user B to selectively set status indication on a percontact basis, to prevent unwanted disclosure. Thus, while theembodiments and associated illustrations present the “minimum set offeatures” they are in no way limited nor restricted to them.

Describing this NANO technology in more detail, conventional VOIPsystems typically preclude users from registering numeric onlyidentities, and in particular numeric telephone identities comprising 10or more digits. Given that the primary identifier for online servicecreation is an email address, many such online services derive andextend their user communities and directories on authenticating usersvia email links.

For example, a user with an email address “ak@starlogik.com” wishing toregister with an online service “ABC dot corn” using the unique(unallocated) screen name “AK”, may receive an email at the aboveaddress containing a link with an authorizing session identifier“12345abcd” (as in “www.abc.com/?a=12345abcd”).

Continuing with this example, since ABC dot com is awaiting activationand confirmation of the link sent to the email address disclosed, whenthe link is activated the service can with high degree of certaintyconclude that the email address belongs to the registering user, sincethe user is required to have authorized access to the email account inorder to activate the service link sent.

While users may know each other by email address, their registered(screen) names are typically unknown. In order to facilitate thelocation of a particular user, many services provide a directory lookupthat can locate registered users by email address, and when located thenreturn the associated “screen name”.

If the registered user has elected to disclose additional personalinformation, for example a telephone number, with the service provider,then it is conceivable to locate the screen name by searching thedirectory on such a “phone number field”. However as described above,searching for a user by entering a telephone number as the registeredscreen name, would typically fail to locate such a user.

Consequently if a VOIP user searches a service directory for a user by aspecifying the telephone number as the screen name, the service wouldtypically return, “user not found”. This directory exception may nowbecome the placeholder for a global new service rule:

“The automatic and systematic creation of the B numbered user account,and the automatic population of the A identities as its onlinecontacts”.

This system generated “numbered account”, the virtual client whosescreen name is now set to their telephone number, is then madeaccessible only to the matching telephone identity calling a serviceGateway, where the calling device identity is now equal to the systemregistered VOIP user account name.

Thus, a single modification to a service registry procedure thatpreviously would preclude users from manually registering a telephonenumber as a VOIP user, and which would consequently return a “user notfound” exception when searching for a screen name comprising a telephonenumber, can now permit conventionally registered VOIP clients tounilaterally add phone numbers (10 digit and longer numeric identities)as system generated and registered VOIP contacts that automaticallyassemble their own contact lists and inherit the presence properties ofconventional VOIP clients.

While it is conceivable to permit users to register online andauthenticate their telephone number as the screen name, thereby“unifying internet and telephony identities” (as disclosed below in onerelated embodiment), such a service is predicated on the user havinginternet access to begin with.

Rather than prescribing Internet access, NANO B clients may be generatedand administered virtually, from the existing VOIP user base A whoidentify them in advance by phone number. As such, the mass telephonemarket B, is registered and captured without any such Internet accessand capability required whatsoever. All that is required for B partyparticipation is a conventional telephone call to a local accessGateway.

Continuing now with FIG. 2:

F2 Step2. Upon entering the B number as described, the VOIP serviceprovider creates a virtual account using the telephone number B as theservice screen name and records Internet user identity A as a contact,thus binding and associating internet user A with telephone number B ina database registry D(40).

F2 Step3. NANO client B is thus systematically created and automaticallyhydrated with net contacts A(1) through A(n), each who have enteredtelephone number B as a contact in their VOIP client.

F2 Step4. Telephone user B dials VOIP access gateway C(20) presentingcaller line identity B at the signaling layer (not shown). In oneembodiment, the call is held in the signaling state, that is the call isnot answered and thus not connected, until it has been ascertained thatat least one associated VOIP user A is online, as per Steps 7 and 8below.

F2 Step5. On receiving the inbound connection from B, Gateway Cinstantiates a virtual NANO client (B#) that serves as the VOIP clientproxy to caller B. That is the B client proxy is automatically spawnedon B connecting to C.

This VOIP proxy is a “server side client application”, programmaticallylaunched and controlled, with the gateway supporting and translatingcommunication protocols between the IP and Legacy telephony domains. Theproxy application is thus logically coupled to Gateway C and mayphysically execute on attendant servers at Gateway C (as shown).

Alternatively the client proxy application may execute remotely at anode connected to Gateway C. In either configuration, Gateway C andProxy B are referenced interchangeably in this disclosure. The B clientproxy instantiation is flow charted in FIG. 8 below.

F2 Step6. B Proxy automatically “signs into the service”, registeringits IP address and port (“IP x:y”) in service registry D(40) permittingother VOIP clients to locate its current address.

F2 Step7. On registering with the service, B proxy receives a list ofits automatically associated net contacts A(n) as described.

F2 Step8. On receiving the contact list, the B client proxy “advertises”its online presence to each contact, using known methods. Onlinepresence information may include the current IP address and port wherethe B proxy is located. Similarly Proxy B receives online statusinformation for each contact A.

On receiving the online presence published, conventional VOIP clientsoftware at A adjusts the B status indicator to reflect “onlinepresence” as described below. Although the advertised presenceillustrated shows unilateral publication from the B proxy to A clients,it is typically a bidirectional process. That is on publishing itspresence to A, B receives online status indication from A in return (notshown).

F2 Step9. Substantially simultaneously to advertising its onlinepresence, B Proxy presents caller B with an interactive session, titled“C menu”. In one embodiment this interactive session plays music on hold(MOH) to B until a VOIP user A originates an inbound call to B (byclicking the now online contact displayed in the VOIP client addresslist).

Since Proxy B has advertised its “on net presence”, MOH allows VOIPusers who have greater call and contact “visibility”, compared with theB telephone audio interface, to drive the connection. Thisadvantageously allows caller B to connect to the Gateway and “await thearrival and announcement” of the incoming and previously rendezvousedconnection from A.

If a single Internet contact A is associated with phone identity B, oralternatively if only one net contact A is online and available, theGateway in one embodiment may announce that default (automaticallyselected) contact by name to the caller using known text to speechmethods, and request the B party to “hold the line”, at which point itis then set to ring the A party presenting B as the calling identity.

By announcing the default contact and by requesting caller B to hold theline, the service presents what may be termed a “secondary caller lineidentity”. With secondary CLI, the service identifies the intendeddestination to B prior to ringing and alerting A, and as such limits theB party exposure. If caller B chooses not to hold the line and therebynot connect through to the default contact A, the destination remainsunaware of the decision to cancel the call.

F2 Step10. On identifying the intended destination A, through a defaultselection as above, else through explicit B selection as detailed below,A and B are logically connected.

F2 Step10 x. If VOIP user A initiates contact by selecting the now“online phone contact B”, the call is routed to the B Proxy client wherethe calling party A is announced to telephone user B, in what is hereintermed a VOIP originated call.

It is important to note, unlike conventional call completion andringing, where a call to telephone B would necessarily terminate andring the device over the legacy PSTN/PLMN network, since the B party isalready connected to the service Gateway and may be located at theregistered Proxy IP endpoint, it is alerted to the incoming call “ONnet”.

That is the VOIP call is routed and terminated at the B servicingGateway and Proxy, rather than conventionally ringing the telephone overthe legacy network. “ON net” call presentation is described below ingreater detail with reference to FIG. 7.

In one embodiment the call announcement at Step10 x may present as a“soft ring”, that is a ring that is played “in channel” directed to thealready connected (ON net) B party, followed by text to speech renditionof the A party user name. In another embodiment, on VOIP user Ainitiating contact to the online telephone B, A may be prompted to speaktheir name and thereby be announced by “a personally recorded soundbite” which is then played to B.

On receiving and accepting the A party announcement, A and B are bridgedand connected and full duplex speech paths are established. If VOIP userA initiates contact and B is presently engaged on an active call withanother VOIP user, standard multi call handling functions may beactivated by the B proxy client and made available to the telephone userB. Such call control functions include automatically placing the activeB call on hold to answer the new A call, and allowing user B toselectively conference active calls.

In another embodiment the interactive session presented to B plays alist of VOIP contacts for selection. If a plurality of Internet contactsA(n) are located (as shown) in one embodiment Gateway C engages caller Bin an interactive session to determine which associated net user Bwishes to establish contact with.

The list of associated VOIP contacts presented (the B party “netextensions”), may be partitioned between contacts that are “online andavailable” and contacts which are “offline and unavailable”. Thisinteractive selection process is described in greater detail below withreference to FIG. 15.

F2 Step10 y. If B makes a net contact selection, Gateway C “invites”destination A by ringing the user with the caller line identitypresented by B. If A answers the call, A and B are bridged and connectedwith full duplex speech paths, in what is herein termed a VOIPterminated call.

To the skilled artisan it will be clear that at Step10 x/y VOIP user Amay be connected and talking to conventional telephone user B, at nocharge. Conversely, telephone user B may be globally connected andtalking to Internet user A, at the local call rate incurred inconnecting to Gateway C.

The artisan will also appreciate the hybrid nature of the resultantcall, where the first communication BC is circuit switched voice, andthe second communication CA is packet switched media.

FIG. 3

Describing the “Tele ON net presence” in greater detail with referenceto FIG. 3, which highlights the Tele IP Presence Protocol fundamentals,seamless tele-presence is delivered to legacy VOIP clients withoutchange.

F3 Step1: Telephone user B(10) dials and presents caller line identity Bto VOIP access Gateway C(20).

F3 Step2: C instantiates a virtual NANO client (B#) that acts as theVOIP client proxy to caller B. Since the Proxy appears to the VOIPcommunity as a conventional client, service properties, including“online presence” are inherited.

F3 Step3: B Proxy registers its telephony IP address and port (IPx:y)with service registry D(40) permitting other VOIP clients to locate it.On registration, B proxy receives a list of previously recorded netcontacts A(n) as described above.

F3 Step4: On receiving its net contact list, the B client proxy“advertises” its online presence to each contact. In the context of thisdisclosure, the term advertising is used to describe the process wherebyone VOIP client notifies another of a change in its online status.

In P2P systems clients typically both notify and receive notification inreturn. In one such system the B Proxy broadcasts its online presence toall its A contacts listed and receives the online status of each Acontact in return.

F3 Step5: In the absence of a telephony presence protocol, conventionalVOIP client software would typically present telephone contacts asgeneric “phone icons”, without any status indication whatsoever.

F3 Step6: On receiving online presence information published by B,conventional VOIP client software may now adjust the B contact statusindicator to reflect “online status” as shown and represented by the“check mark”.

Certain state indicators may be automatically selected and presented,for example, if caller B is actively engaged on a call with anotheruser, the client proxy may publish an “on call” status. For simplicity,the default status described and indicated in the embodiments is “ON/OFFnet”.

It will be clear to the skilled artisan that more “granular onlinepresence” may be indicated by the B Proxy client, for example bypermitting phone B to select amongst a plurality of online statusindicators, including common states such as “available”, “invisible”,“busy” and “Do Not Disturb”.

While this illustration presents the NANO client “screen name” as thetelephone number, it is common for VOIP client address books to allowusers to enter a more recognizable “username”, which is then displayedin conjunction with or in place of the screen name.

For example, assuming VOIP client A enters “AK” as the username for theNANO “14154125111” screen name, the displayed contact in the panelsdepicted in Steps 5 and 6 could then read “AK 14154125111” or simply“AK”.

The numeric screen name presentation in the embodiments has been adoptedto avoid confusion between the newly disclosed and system generated“telephone number screen name” and the conventional “alphanumeric screenname”.

This also serves to highlight that a telephone contact may now bepresented as a “conventional VOIP client” albeit it a purely numericalone.

FIG. 4

Continuing with Tele Presence as disclosed above and describing now amore personalized function with reference to FIG. 4, which illustratesTele IP Presence Protocol Aliased.

F4 Step1. VOIP user A(30) enters the telephone number B(10) of anexisting and conventional VOIP contact with screen name Ab, therebyaliasing the resultant NANO B client as net contact Ab. Similarly, anyconventional VOIP contact “An” may have their telephone number thusassociated.

Since NANO client B has “owner alias Ab” information recorded, thisonline presence may indicate to user A that “Ab” rather than simply “B”is now online. Similarly if B requests connection to A, named calleridentity (Ab) may be presented rather than just B (“number”) lineidentity.

F4 Step2. On entering the telephone number B of contact Ab, serviceregistry D creates or updates the B NANO client address book byrecording both “A as the net contact” and “Ab as the net owner alias”.

F4 Step3. The B client now has both the net contact A and the B screenname alias whose telephone number was entered by A. This telephonenumber is unauthenticated, since A is the disclosing party. However in arelated embodiment, owner authenticated phone numbers may beautomatically propagated to users online address books, therebyautomatically and securely populating the telephone number field of acontact.

While it is typical for online contact address books to support anynumber of related information fields, of which a phone number may beone, these collateral contact fields are “passive” in nature. That isthey do not display nor inherit any VOIP client functionality.

In one embodiment, the aliased system permits VOIP user A to record asingle unified “VOIP and NANO telephony contact”, by adding a telephonenumber to an existing VOIP address book entry. However unlike a passivephone number field, the system automatically “activates and transforms”the embedded telephone number field into a NANO virtual client asdescribed.

As such when the telephone and its associated NANO client registers as“ON net”, rather than separately indicating and identifying “B onlinepresence”, the online status indicator for the aliased contact isupdated, thereby presenting a consolidated status and single view of thecontact.

As described in FIG. 3 above, when B dials and connects to Gateway C, itinstantiates and registers the B Proxy client with service registryD(40). On registration, B proxy receives a list of previously recordednet contacts A(12 n).

F4 Step4. On receiving the contact list, the B client proxy “advertises”its online presence to each contact. However in addition to publishingthe B client details, the aliased system and method now publishes both Band the alias data set. This permits net contact A to update the statusof the alias.

F4 Step5. By example, a conventional VOIP contact with screen name“babybel”, which also has a telephone number 14154125111 (abbreviated “. . . 5111”) is shown in “OFF line” status as indicated by the “X” icon.

F4 Step6. Continuing with this example, on receiving online presenceinformation published by Proxy B, VOIP A client software may now adjustthe VOIP contact “babybel” to reflect “online status”, as shown by the“check mark”. While a conventional online icon is displayed, an icondepicting “tele-presence” (that is the contact is online via phone) maybe indicated to differentiate from “conventional presence” (where thecontact is connected via PC).

FIG. 5

With reference to FIG. 5, Sample Stored Procedures transforming contactsfrom conventional telephone numbers into NANO virtual numbered clientsis flowcharted.

The left panel charts a typical stored database procedure that executeson updating contacts (for example inserting a new, changing an existing)in a registry.

The right panel charts a typical procedure that automatically executeson selecting previously recorded contact information from a serviceregistry.

The latter procedure permits the VOIP service provider to automaticallyconvert all conventionally recorded telephone number contacts that arealready in the system to the new NANO client format as disclosed withoutuser intervention.

Describing the Left Panel:

F5 Label ˜100: User A enters phone number B as a contact screen name.

F5 Label ˜110: Service checks whether a B#client exists in the serviceregistry.

F5 Label ˜110: [No] IF no such B#client exists logic flows to ˜120

F5 Label ˜110: [Yes] IF the B#client exists logic flows to ˜130

F5 Label ˜120: Service creates a new VOIP client with B# as the screenname in the service registry.

F5 Label ˜130: Service records user A as a VOIP contact in the B clientcontact list. B now has A as a net contact.

With reference to the Right Panel:

F5 Label ˜101: User A signs into the VOIP service.

F5 Label ˜102: On authenticating A (not shown) the service retrieves allrecorded contacts from a service registry.

F5 Label ˜103: Service determines if any regular phone number contactsexist in the returned contact data set. If so logic flows to ˜104otherwise regular service login continues (not shown)

F5 Label ˜104: Service automatically converts each conventionallyrecorded telephone contact into the corresponding B#client as describedper ˜110 in the Right Panel above.

FIG. 6

Describing now with reference to FIG. 6, a VOIP originated call flow tophones OFF net, being the state assigned to phones that are disconnectedfrom the service gateway.

F6 Step1. Internet user A(30) dials telephone number B(10). The call isswitched and routed toward Gateway C(20) interconnecting to the PSTN. Cis an “edge gateway” that is local to the B destination network.

F6 Step2. In one embodiment, VOIP client A is aware of the B phonestatus and location as advertised and disclosed in the Tele IP Presenceprotocol described in FIG. 3. As such, call routing logic may bedetermined at the client.

In another embodiment, a VOIP proxy server (not shown) determines whichgateway to route the connection toward, based on the country and in someinstances the network code in the B number. Such a proxy server nowqueries registry D(40) to determine whether the NANO B client proxy isactive and registered, as described earlier in FIG. 2.

In a further embodiment Gateway C, on receiving the inbound connectionfrom A, performs the query to determine whether the NANO B client proxyis active and registered. Regardless of which network node determinesthe availability of the B client proxy, since in this example telephoneB is not connected to C, it has no active proxy and IP endpoint in D andis thus determined to be “OFF net”.

F6 Step3. On determining that telephone B is “OFF net”, that is it isdisconnected from gateway C, the network node currently processing theconnection request from A, redirects the call to an Internet hostedmessaging platform, an IVR (Interactive Voice Response) system as shown,thereby terminating the connection on the Internet.

F6 Step4. On caller A selecting a messaging feature directed towarddestination B (as detailed in FIG. 10 below), telephone B is notified.Messages deposited by A destined for telephone B, are stored in amailbox automatically created for B on the Internet platform.Notifications sent to B may include the actual message, or may include alinked reference to the message, the content of which is retrieved whenB connects to the service by activating the link.

The advanced “ON/OFF net” device state determination and resultantswitching for VOIP originated calls addressed to legacy telephones,uniquely ensures that all such Internet originated calls terminate onthe Internet, and consequently remain within “the free service domain”.

While nominal cost may be attracted in delivering notification messages,there is no associated “clocked terminating penalty”, no air timeconsumption and billing, as all circuitry required to connect when B is“ON net” and to record and deposit a voice message when B is “OFF net”,is established on the Internet, rather than over legacy networks as isrequired with conventional switching methods.

FIG. 7

Continuing with Internet call routing, FIG. 7 describes a VOIPoriginated call flow to phones “ON net”, being the state assigned tophones that are connected to the service gateway.

F7 Step1. Telephone B(10) dials and connects to Gateway C(20),instantiating a B client proxy that registers its IP address and Portwith registry D(40) as described above in FIG. 2.

F7 Step2. Internet user A(30) dials telephone number B(10). The call isswitched and routed toward Gateway C(20) interconnecting to the PSTN.

F7 Step3. In one embodiment, VOIP client A is aware of the B phonestatus and location as advertised and disclosed in the Tele IP Presenceprotocol described in FIG. 3. As such, call routing logic may bedetermined at the client.

In another embodiment, a VOIP proxy server (not shown) determines whichgateway to route the connection toward, based on the country and in someinstances the network code in the B number. Such a proxy server nowqueries registry D(40) to determine whether the NANO B client proxy isactive and registered, as described earlier in FIG. 2.

In a further embodiment Gateway C, on receiving the inbound connectionfrom A, performs the query to determine whether the NANO B client proxyis active and registered. Regardless of which network node determinesthe availability of the B client proxy, in this example telephone B hasalready connected to C, it has an active proxy and IP endpointregistered in D and is thus determined to be “ON net”.

F7 Step4. On determining that telephone B is “ON net”, that is,connected to gateway C, the network node currently processing theconnection request from A routes the call to the B client proxy at C,terminating the connection on the Internet.

F7 Step5. On terminating the call at the B client proxy, the callingparty A is announced to telephone user B as described in FIG. 2 Step 10x above. The VOIP originated call is thus routed and terminated at the Bservicing Gateway and Proxy, rather than conventionally ringing thetelephone over the legacy network.

F7 Step6. On accepting the announced call A and B are bridged. Thetelephony bridge is constructed by merging the RTP media streamestablished on the call path between A and C (“the internet leg”) andthe TDM voice circuit between B and C (“the conventional leg”). Thebridge performs any protocol conversions between the telephony domains.

FIG. 8

The disclosed telephony IP presence protocol that governs the “ON/OFFnet” caller state is in the affirmative, the “ON net” state, contingenton the establishment of the NANO client proxy. FIG. 8 flow charts theVOIP virtual client proxy instantiation.

F8 Label ˜100: Telephone user B(10) dials Gateway C(20).

Prior to dialing Gateway C, phone B is in an “OFF net” and disconnectedstate, depicted by Graphic ˜101 (solid white filled elliptical, enclosedblack phone). The “OFF net” state is the default state, determinant inthe absence of the VOIP proxy and resultant IP address registration.

F8 Label ˜200: On call setup, C receives calling line identityinformation (B#) associated with device B. C then programmaticallyspawns a virtual VOIP client that signs into the VOIP service using B#as the service screen name. The Client Proxy is depicted by Graphic ˜201(dotted white elliptical, enclosed black PC).

As mentioned earlier in the embodiments, this VOIP proxy is a “serverside” client application, programmatically launched and controlled, withthe gateway supporting and translating communication protocols betweenthe IP and Legacy telephony domains.

The proxy application is thus logically coupled to Gateway C, and mayphysically execute on attendant servers at Gateway C. Alternatively theclient proxy application may execute remotely at a node connected toGateway C. In either configuration, Gateway C and Proxy B are referencedinterchangeably.

In one embodiment the VOIP client application thus launched uses adefault system password coupled to the B#screen name when signing intothe service. In another embodiment the system may request a passwordfrom the caller prior to the Proxy signing into the service.

During this client proxy instantiation and service registration, B mayremain in a logically connected state to C. That is, B may be connectedto C via a call setup and thus signaling channel, until such time as thecall is explicitly answered and accepted by C. Alternatively, B mayphysically connect to C, that is C may answer the call establishing avoice circuit between B and C.

One advantage to holding the connection in the signaling state is thatcaller B is not charged until the client registration is complete. Inone embodiment, B may hear an extended “ringing” and remain in thisquasi connected state until at least one associated VOIP net user A isavailable and online to receive a call, as described earlier.

In a further embodiment, B is only physically connected when a VOIP userA requests a connection to B, as per FIG. 2 Step10 x. In one such anembodiment, B would hear an extended primary ring, while connecting tothe gateway, followed by a secondary “ring” on the gateway answering B,to indicate an inbound connection has presented, followed by the callerA announcement as described.

F8 Label ˜300: On signing into the VOIP system, the VOIP client proxyregisters its IP address and Port with registry D(40). This Internetaddress describes the VOIP endpoint to which calls may be directed. TheIP address is depicted by Graphic ˜301 (dotted elliptical, enclosedIPV4).

By example, one such virtual client proxy registration may be recorded,and subsequently located, as user “14154125111@sky.com”, where“14154125111” is the B calling line identity and “sky.com” the VOIPservice provider realm. Dialing this Virtual VOIP client would thenresolve to the IPV4 address “192.10.30.60” and port “1024” endpoint asdepicted.

F8 Label —400: When the B client proxy registration is complete, B is“ON net” as depicted by Graphic ˜401 (solid black filled elliptical,enclosed white phone) and by the conventional VOIP client online statusindicator (“ticked”). B telephony presence is then advertised toassociated VOIP users A as described in FIG. 2 above.

FIG. 9

The IP telephony presence protocol described, controls IP originatedcall routing described now with reference to FIG. 9, which presents theVOIP ON/OFF net switching flow chart.

F9 Label ˜100: VOIP user A dials telephone user B.

F9 Label ˜200: During call processing the VOIP system determines whetherphone B is ON/OFF net.

In one embodiment this determination is made on assembling the NANO Bclient service name, by example “14154125111@sky.net”, to which theconnection request is addressed. On addressing the connection thus, theservice then resolves the user name into the current location of the Bproxy client by querying a service registry (not shown).

F9 Label ˜200: [No] If the service registry does not contain an activeand current address location for the B proxy client, B is OFF net andlogic then flows to ˜300.

F9 Label ˜200: [Yes] If the service registry does contain an active andcurrent address location for the B proxy client, B is ON net and logicthen flows to ˜400.

F9 Label ˜300: Caller A is routed to an Internet hosted InteractiveVoice Response (IVR) messaging platform. The messaging functionsprovided by this platform are described in greater detail in FIG. 10following.

F9 Label ˜400: Caller A is routed to the IP address and Port where the Bclient proxy is located at Gateway C.

F9 Label ˜500: On receiving the incoming call request from A, Gateway Cin one embodiment alerts the connected phone B by presenting a shortsecondary ring (“the soft ring”) “in channel” and announcing the A useridentity, as described earlier in FIG. 2 Step10 x above.

F9 Label ˜600: On presenting the call from user A, gateway C awaitsacceptance from user B. In one embodiment B may be requested to press apredetermined key in order to accept the call. In another embodiment Bmay accept the call on issuing a predetermined voice command. In oneembodiment B may decline to accept the call by failing to provide anyresponse or input to the announcement.

F9 Label ˜600: [No] If B does not accept the call, the connection isrerouted to the Internet messaging platform as per ˜300 above

F9 Label ˜700: [Yes] If 13 accepts the call, C places any active B callon hold and connects A as described earlier in FIG. 2 Step10 x above.

FIG. 10

Continuing with IP call routing in FIG. 9 Label ˜300, the InternetMessaging functions presented to VOIP user A calling phone user B, whois either OFF net or declines to accept the call are now described withreference to FIG. 10, which illustrates delivering OFF net callnotifications.

F10 Label ˜300: VOIP user A is presented with a messaging menu directedtowards phone user B (the “AB” menu as depicted).

In one such service menu, A is prompted to press “l” to send a call backmessage, “2” to leave a voice message recording and to “disconnect” inorder to deposit a “missed call” notification.

F9 Label ˜310: On selecting option “I”, the messaging platformautomatically constructs and sends a “callme” text to phone B.

F9 Label ˜311: One such callme text message delivered to B, includes theA Internet user identity (“AK@sky.com”) in addition to the local returngateway access phone number (“73132421581”).

F9 Label ˜320: On selecting option “2”, the messaging platform permits Ato record a voice message for B. This voice message is then deposited ina “B numbered account” automatically created on the Internet Platform,with user A identity information recorded as the sending party.

F9 Label ˜321: One such voice message notification delivered to B,includes the A Internet user identity (“AK@sky.com”) in addition to thelocal return gateway access phone number (“73132421581”).

F9 Label ˜330: On selecting neither option presented, in one embodimentcaller A may disconnect the call (as depicted by “X”) on reaching theMessaging Platform to deposit a missed call indicator on device B.

F9 Label ˜331: One such missed call notification is delivered by Cautomatically connecting and disconnecting an outbound call to B, withcaller line identity set to the local return gateway access phone numberas shown. In such a notification delivery, C immediately disconnects thecall “on ring back tone presentation”, on receiving notification thatthe B party device is ringing, in order to prevent the call beinganswered.

On phone user B selecting the gateway access phone number and returningthe call, the connection is routed back to now associated VOIP user A.If any voicemail messages are awaiting B retrieval, these messages maybe accessed from the Internet Messaging Mailbox identified by the Bcalling line identity and played back to B, prior to optionallyconnecting the call to user A.

FIG. 11

While the “AB” user associations described in the above embodiments arerecorded at the formative stage, during a contact capture phase (F1Step1), they may similarly and logically be recorded later during aconventional call establishment cycle. In particular associations may berecorded directly at the Gateway servicing a conventionally switchedVOIP A originated call to phone B that is terminated over the PSTN.

In one such a “late binding” embodiment and with reference now to FIG.11, which depicts abstracted core service elements in a dial streamembodiment, the method and system disclosed hooks “directly into theVOIP dial originating stream”, by automatically recording the logical ABrelationship in an edge cached service registry, as and when aconventional VOIP call to a telephone presents at an IP Gateway to thePSTN servicing the dialed destination region.

F11 Step1. VOIP user A dials phone B and connection is routed to GatewayC on what is a conventionally VOIP originated and PSTN terminated call.

F11 Step2. C receives the Internet outbound call originated from A totelephone B, and records the BA user association in local registry D.

F11 Step3. C establishes the forward connection towards B, ringing andterminating the connection on device B over the PSTN.

In one Gateway embodiment as described in the figures above, theconnection between a VOIP user and a telephone user is formed bybridging a first and second call leg, AC and CB respectively, wheretypically B rings showing the C access number as calling party identity.

F11 Step4. Given the advanced automatic association between A and Bduring the forward call establishment, B may now at any time return callC and connect back to A.

This B return call may follow various A connection request scenarios,including:

(i) returning a missed call (call from A going unanswered and phone Bdisplaying a “missed call from C”)

(ii) replying to a voicemail message from A (the call having goneunanswered and caller A deposits a message)

(iii) returning a call after a conventionally established conversationhas completed (A and B successfully connected).

F11 Step5. On B returning the call to access number C, the Gatewayqueries registry D to determine any VOIP users A associated with thecalling identity B.

F11 Step6. On B selecting the desired associated contact A, as describedabove, B is connected through to A on a reverse (VOIP terminated) callpath.

This automatic “just in time association between net user A and phoneuser B”, recorded directly in the call signaling path and circuitry atthe edge Gateway, permits any call originating from any internet user Ato present “generic” gateway calling identity to telephone user B, andyet still permit B to return the call back to the originating party A,using the reverse caller identity and contact selection methods asdescribed.

FIG. 12

Continuing with one such “dial stream association” embodiment, FIG. 12shows a basic VOIP originated dial stream stepladder that will beevident to the skilled switching artisan.

F12 Step10. VOIP user A(1) dials phone B(6).

F12 Step11. Connection is routed via optional outbound service Proxy(2)to PSTN gateway C(3) on what is a conventionally VOIP originated andPSTN terminated call.

F12 Step12. C receives the Internet outbound call originated from A totelephone B, and records the “BA user association” in registry D(4).This BA dataset includes the “B#” (the called party phone number) andthe “A@” (the Internet caller identity).

The D record is thus indexed on B# as the primary access key in order tologically group all VOIP callers A contacting the same B, and tofacilitate later retrieval, when B returns via C. D may be logically andphysically the same node as C. Alternately D may be distinct from C(connected as shown).

F12 Step13. C establishes the forward connection towards B by typicallysending an IAM (Initial Address Message) directed to destinationSwitch(5) servicing the telephone subscriber. The IAM message headerrecords the phone number describing gateway C as the calling lineidentity.

F12 Step14. On receiving the IAM the switch forwards a call setupmessage to the B device, which in turn responds with an “alerting”message back to the switch.

F12 Step15. On receiving the “alerting” message, the switch returns anACM (Address Complete Message) back to gateway C advising that the Bdevice is ringing. B now rings displaying gateway C as calling lineidentity.

F12 Step16. On receiving the ACM, gateway C presents the ring back toneto caller A.

F12 Step17. Caller A hears phone B ringing.

Thus with the single additional “Step12”, conventional callestablishment between VOIP and PSTN domains now capture and establish a“reverse call path”, accessible when calling from PSTN to VOIP, on theautomatic BA association recorded along the forward path as disclosed.

Over time, this automatic reverse address association may result in aplurality of Internet Users A(n) being associated with a singular phoneuser B. This “BA(n)” recorded data set, representing the “N” distinctVOIP users who have called phone user B, thus forms the “B netextensions” presented when B calls C.

This VOIP A contact list may be presented to phone user B in a varietyof ways, as disclosed, including presentation as a time sorted LIFO(Last In First Out) list, which presents “the last A contact first”,permitting B to “hold the line” in order to connect to the most recentcaller without explicit selection.

In a LIFO presentation, B may activate a key to select from theremaining list of earlier contacts. In another presentation, the contactlist may be partitioned into VOIP users who are online and available,and users who are offline and consequently unavailable to accept areturn call at the time B calls C.

FIG. 13

Describing a late binding call completion embodiment in greater detailwith reference to FIG. 13, which shows a basic VOIP redirection callflow chart, A is redirected to an intermediary Internet MessagingPlatform when B is OFF net, unavailable or when the call goes unansweredor is declined.

In one embodiment, on ringing B with generic gateway C as the primarycaller identity, the connection between A and B may be asymmetricallyestablished to deliver a secondary caller line identity feature towardsparty B, whereupon B answering the call, C may announce the A netidentity prior to completely connecting the A party through.

F13 Label ˜100: A dials B.

F13 Label ˜200: The call is conventionally switched and routed over theInternet to PSTN gateway C, which rings telephone B.

To the skilled artisan it will be evident that the call between A and Bmay be logically and physically established by bridging a first andsecond leg, being the originating leg from A to gateway C and theterminating leg from gateway C to B (“AC and CB” respectively), with Cperforming the requisite protocol, and where necessary, codecconversions between the Internet and the Legacy network.

As such, call connection progression and control between A and B may beasynchronously established and separately controlled by theinterconnecting gateway and attendant call processes.

F13 Label ˜210: In a preparatory step, C records the BA addressinginformation in Registry D (not shown) for future reference (when B goes“ON net” by connecting back to C).

F13 Label ˜300: [No]. IF B is unavailable or does not answer, logicflows to ˜400.

F13 Label ˜300: [Yes]. IF B answers the call, logic flows to ˜500

F13 Label ˜400: On B failing to answer, the call is automaticallyredirected to an Internet hosted messaging platform, permitting A todeposit a message for B without paying. That is, caller A is switchedback onto the Internet to avoid the conventional terminating penaltyincurred on switching unanswered calls to legacy messaging platforms.

The messaging platform may present an interactive session to A, where inone embodiment a plurality of messaging options is provided, resultingin the automatic creation of a B party messaging inbox on the Internet.Sample messaging logic is depicted in the dotted and minimized call flowpanel that references the expanded FIG. 10 above.

In one such interactive session A may select between leaving a voicemailmessage (for example, by pressing “1” or holding the line) and sending a“callme” text notification (for example, pressing “2”).

F13 Label ˜500: IF B answers the call, C announces the A Internet useridentity, delivering what may be termed a “secondary caller lineidentification”, in that the primary calling line identity that ringstelephone device B may be set to the generic Gateway C access phonenumber.

In one embodiment, while C announces the VOIP user identity to B (forexample using “text to speech” rendering of the A user “screen name”)the A party continues to hear the ring back tone.

That is, A hears telephone B ringing, while B is being informed of thenet caller identity “in band, in the CB call path”. B may thusselectively decline the incoming call, for example by disconnecting,without undue exposure to A.

F13 Label ˜600: IF B disconnects or declines A, the AC call leg isredirected to the Internet messaging platform and call processingproceeds as per ˜300 above. This call redirection may beprogrammatically delayed, to veil the fact that B explicitly declined toaccept the call, so as to appear to A that the call went unanswered.

F13 Label ˜700: IF B does not disconnect after the net user identityannouncement, C connects A and bridges the call to B to deliver fullduplex speech paths between A and B resulting in conventional telephoneconversation.

FIG. 14

The automatic net and phone user “address association” permitsconventional telephony users to return calls to Internet users.Describing now one such return call originated from phone B, via GatewayC, to an automatically associated VOIP user A as described above, FIG.14 depicts a basic VOIP terminated switching stepladder.

F14 Step10. Phone B(1) dials and connects to gateway C(3) routed via anoriginating Switch (2).

F14 Step11. C queries service registry (4) on B calling line identityreturning, the identity of previously associated VOIP user A.

F14 Step12. C announces the associated VOIP A user identity to caller Bon establishing a first voice circuit (“leg1”) between B and C. Ifmultiple A were previously associated with B, selection methods arepresented as disclosed (not shown).

F14 Step13. On selecting an associated user A, where in this illustratedexample caller B “holds the line” and thereby selects the defaultannounced user A, C invites a connection between “B and A”, setting the“From” contact header data as originating from phone number B.

F14 Step14. During connection invitation, service proxy (5) resolves thecurrent location of VOIP user A, assigns a media channel back to gatewayC and rings destination user A.

F14 Step15. On ringing A, system progress messages (“183 ringing”) flowback to gateway C, informing the gateway that the destination isringing. C presents the ring back tone to B.

F14 Step16. B hears A ringing.

Fl4 Step17. When A answers the call, a second media circuit (“leg2”) isestablished between A and C. At this stage, C may apply one of variousconnection and bridging architectures, to create voice paths between Aand B.

To the skilled artisan it will thus be evident that the call pathbetween phone user B and VOIP user A may be logically and physicallyestablished by bridging a first and second leg, an originating leg fromB to C and a terminating leg from C to A (“BC and CA” respectively),with C performing the requisite protocol, and where necessary, codecconversions between the Legacy and IP network.

As such, call connection progression and control between the B and Aendpoints may be asynchronously established and separately administered.In one such bridging architecture, C may bridge the first and secondcall legs (BC and CA), transporting media flow between the two. Inanother architecture, C may extricate itself from the media path,bridging and passing media directly between A and B, peer to peer,whilst maintaining call signaling and process control.

FIG. 15

By introducing a “time dimension” to return call user selection,frictionless callback may be delivered from legacy to VOIP networks,permitting the phone user to “hold the line” for the most recent netcontact. FIG. 15 illustrates this timed VOIP terminated switching.

F15 Step1. VOIP user A(30) dials phone user B(10).

F15 Step2. Gateway C(20) automatically records user A in a B clientreferenced contact database. In addition to recording the contactidentity, C stamps the A record with the connection request date andtime (T1).

F15 Step3. This automatic contact management system creates and updatesthe B phone user record, listing net A user together with connectiondate and time (A1 and T1).

F15 Step4. Per the Tele IP Presence Protocol disclosed, in theillustrated embodiment C determines that B is “OFF net” andautomatically disconnects caller A.

F15 Step5. C notifies B that a connection was requested, in theillustrated embodiment by ringing and automatically disconnecting B withcaller identity set to C. This to deposits a “missed call” on B withorigin set to C (as depicted by the “C” labeled semi circle arrow).

Steps 4 and 5 above describe an alternate “OFF net” switching embodimentthat applies a “default notification”. In this embodiment Cautomatically “disconnects caller A and pings destination B”, ratherthan redirecting A to an Internet hosted messaging platform forinteraction as described earlier.

F15 Step1 x. Over time additional VOIP users A(2 . . . n) may dial phoneB.

F15 Step2 x. For each user A that dials B, C automatically records andtime stamps the connection request in the B indexed contact database.

F15 Step3 x. On receiving multiple connection requests from multipleVOIP users, the B contact database records a LIFO (Last In First Out)list of users A(n . . . 2,1) reverse sorted on times T(n . . . 2,1).

F15 Step6. B dials C, returning the missed call(s) to establish a firstcommunication path “BC”.

F15 Step7. C queries the contact database using the B calling lineidentity as access key and returns with the LIFO A(n . . . 2,1) userlist. In this list “A(n)” represents the most recent VOIP user thatrequested contact with B.

F15 Step8. C announces the “default selected net contact” user A(n)identity to B. In one embodiment this announcement is made using a “textto speech” rendition of the VOIP A username. This announcement deliversa “secondary caller line identity” feature towards B.

C may announce additional selection options (not shown) that may permitB to select inter alia, earlier contacts A(n−1 . . . 2,1) and contactsthat are “online and available”. B may disconnect after hearing thedefault selected (most recent) contact announcement without undueexposure to the announced user.

F15 Step9. B “holds the line” in order to connect to the announced userA(n). This implicit contact selection causes C to ring user A(n) withcalling line identity set to B, establishing a second communication path“CA”.

F15 Step10. IF A answers the call, C bridges the first and secondcommunication paths (BC and CA) to create voice circuitry between A andB.

FIG. 16

Continuing with the LIFO contact management system described above, FIG.16 illustrates a sample Menu Play list flow chart.

F16 Step1. Phone user B(10) dials Gateway C(20).

C queries the contact database using the B calling line identity asaccess key and returns with the LIFO A(n . . . 2,1) user list asdescribed in FIG. 15 above. In this list “A(n)” represents the mostrecent VOIP user that requested contact with B.

F16 Step2. C plays a contact selection menu to B as depicted by the boldrectangular speech bubble (“≡”). One such menu, detailed in the expandedflow chart intersecting with the bubble, presents as follows:

F16 Label ˜200: C announces the most recent VOIP user contact A(n) in amenu that presents three basic contact navigation options, instructing Bto:

(i) press “0” or “hold the line” to ring the announced user A(n)

(ii) press “#” to skip the current and present the next most recentA(n−1)

(iii) press “*” to present an enumerated (alphabetical) user list A(12n)

F16 Label ˜300: C awaits selection from B. Selection may be indicatedusing standard DTMF signals corresponding to keys pressed on thetelephone device. Alternatively selection may be indicated by spokenvoice commands. After a predetermined delay, if no selection isindicated (B “holds the line”) C rings the announced user A(n).

F16 Label ˜400: IF B presses the pound (#) key logic flows to ˜410

F16 Label ˜500: IF B presses the star (*) key logic flows to ˜510

F16 Label ˜600: IF B presses the zero (0) key logic flows to ˜610

F16 Label ˜410: On receiving the #signal, C skips the current user A(n)and presents the next most recent user A(n−1) on menu logic that flowsback up to ˜200 along path ˜420.

F16 Label ˜510: On receiving the * signal, C presents an enumerated userlist A(12 n) and awaits selection (not shown). One such list may supporta plurality of selection methods, including numeric selection (matchingthe enumerated contact presented), “abc/2” encoded spelling (pressingkeys corresponding to letters that match the user name or part of theuser name), voice matching (speaking the user name) and so on.

F16 Label ˜610: On receiving the 0 signal, alternatively after apredetermined interval where no signal is received, C rings theannounced user A(n).

In a further LIFO presentation embodiment, VOIP clients may be permittedto “touch B”, that is to select the B contact without actuallyrequesting a connection, and thereby “prime the service registry andgateway” by transmitting a system generated timestamp to reorder thecontact list. Priming the gateway thus, programmatically “bubbles theselected contact up to the front of the list”, establishing it as themost recently “touched” contact, and thereby instructing the gateway topresent the primed contact as the default selected contact.

In another presentation embodiment, the associated A client list may bepresented via well known protocols, including text to speech. In such arendition, the VOIP username is read back to the caller using eithersynthesized or natural language composition. Further, service shortcutsmay permit the B caller to key the digits corresponding to the lettersthat comprise a username, to “jump to and select” a known previouscontact.

Client selection may also be made using voice recognition and patternmatching algorithms, whereby B “speaks the name of the desired Acontact” and if the spoken name is matched to a contact, the contact isthen selected. Voice recognition in this instance can provide resultsgiven the limited dataset, the relatively small number of A clients whohave associated themselves with the B caller.

FIG. 17

In one embodiment, delivering the automatic contact association betweenVOIP users A and phone B, and the resultant reverse switching on callback, may be enabled on a new “virtual service realm.” One suchembodiment permits users to address conventional telephone connectionson a new outbound “SIP domain” as per FIG. 17, which illustrates theVirtual Realm service abstraction.

By abstracting conventional telephones on a “well known address”(“dom.com” in the illustrated example), any existing VOIP client may bepermitted to address any conventional telephone as an “IP telephone” andto be serviced according to the newly disclosed methods.

The Virtual Realm disclosed uniquely combines and leverages two InternetRegistry Protocols, namely Domain Name Systems (DNS) and ENUM (IETF RFC3761) to deliver a highly distributed data switching architecture,geographically mapped over the E164 dial plan. The resultant DDNS(Dynamic DNS) is outlined here, expanded in FIG. 18, and described inyet greater detail later, with reference to figures in appendix B.

The DNS “query response system”, well understood by skilled artisans, istypically used to resolve service names into IP addresses. As the ManMachine Interface (MMI) to the Internet, DNS permits services to beabstracted and accessed using “names” (for example “www.dom.com”) ratherthan locating them directly on the underlying Internet Protocol “machineaddresses” (for example, an IPV4 address “196.010.030.060”).

DNS records mapping to Internet services are overwhelming “static” inthat the association between host name and underlying IP address changesinfrequently. This static association permits DNS records to be cachedand distributed out to nodes closer to the querying peripheral. DNS useswell known Authoritative Root Servers and Zone Delegation, together withregister attributes such as TTL (Time To Live), to ensure that addressremain current and changes are propagated and cached as and whenrequired.

DNS is a highly scalable and proven IP switching service that delivers anear real time “query response” when resolving names into theirunderlying IP addresses. Since most DNS resolutions change infrequently(for example, web and email host addresses), the worldwide propagationand caching mechanism permits local routers to cache the resultant IPaddress in order to deliver a more distributed and responsiveresolution.

Dynamic DNS (DDNS) is an advance that permits “late bindings” betweennames and their associated IP address, popularized by home basednetworks and servers. These servers typically operate on limited privateJP address ranges that are dynamically mapped to shared publicaddresses, a method that extends and, to some extent, overcomes thescarcity of available addresses in the IPV4 (32 bit) address space.

DDNS works on the principle that L3 domain IP addresses (for example,the “new” in “new.dom.com”) are constantly changing. Since the delegatedL2 domain (the “dom” in “dom.com”) is typically fully propagated andthus “known”, programmatically modifying L3, to which L2 is theanswering authority, permits instant resolution and close to zeropropagation time for registering new services.

In one embodiment, DDNS is thus an advance that permits instant L3domain registration and resolution that extends the domain space to theleft of the principal (“dom.com” for example) without incurringadditional registration fees, as payment only applies when acquiring theL2 domain. L3 is a virtual, private and unrestricted name space, knownas the “wildcard” (star) domain.

In addition to supporting dynamically assigned IP address resolutionapplied to Gateways and Media Servers located in the wildcard domain,the methods and systems following extend DDNS application to incorporateENUM, as applied to telephone numbers, thereby permitting an unlimitednumber of L3 and greater domains to be dynamically added to the DNStree.

Since these star domains are “authoritative” (TTL near zero) they haveclose to zero propagation time. Further, since ENUM permits zonedelegation at every point (digit) along the dotted path, a dynamic andmassively distributed data switching network is delivered,geographically mapped on the E164 numbering plan and switched over theconventional DNS protocol.

E164 Zones form the following international country code dialing matrix:

Zone 1: North American Numbering Plan

Zone 2: Africa (mostly)

Zone 3: Europe

Zone 4: Europe

Zone 5: Latin America (mostly)

Zone 6: South East Asia and Oceania

Zone 7: Seventh World Numbering Plan (former Soviet Union)

Zone 8: East Asia and Specialized Services

Zone 9: Central South and Western Asia

Zone 0: Unassigned

Describing now the Virtual Realm fundamentals:

F17 Step1. VOIP user A, by example, registers with username “a@sip.net”.This registration is facilitated by a Registrar Server and recorded in aLocation Server (both not shown). The data recorded in the LocationServer maps the public address of the user (the “Address Of Record”) tothe current IP address of the SIP phone.

F17 Step2. User A, who wishes to establish communication with telephoneuser B, enters the phone number (“14154125111”) followed by the “VirtualRealm” (the new service domain) that together describes a conventionalSIP URI (“14154125111@dom.com”).

F17 Step3. In one embodiment the client software invites the connectionrequest via the Inbound Proxy Server P, described by the virtual domain(“dom.com”). The address of this proxy server may be determined usingconventional DNS SRV resolution. Alternatively the Inbound Proxy may belocated by querying appropriately configured DNS SRV records.

On receiving the connection request, the Inbound Proxy rewrites theinitial destination URI “14154125111@dom.com” to now support thedisclosed DDNS ENUM distributed architecture, resulting in a newdestination URI “14154125111@1.1.1.5.2.1.4.5.1.4.1.dom.com”, thusrelaying the connection to a “1 dot” US zoned and delegated Gateway Cwith associated Registry D. ENUM reverse dots the telephony domain sinceDNS is parsed “right to left”.

It will be evident to the skilled artisan, that this ENUM URI permitsthe virtual service realm to be zoned and delegated anywhere along thedotted path to the left of the primary domain (“dom.com”). This permits“edge caching”, to record, the VOIP user A and telephone user Bassociation and to deploy the new switching logic directly at the “edgegateway” to the legacy PSTN.

In the illustrated example, ENUM DNS is at minimum delegated on “Level3” (“1.dom.com”) pointing to a gateway servicing the USA region, asdefined by the leading “1” Country Code per the ITU E164 geographicnumbering plan.

F17 Step4. On relaying the connection to the zoned and delegated URI asdescribed, gateway C receives the connection request, where data in theconnection request headers include the “From” (A=a@sip.net) and “To”(B=14154125111) fields. C extracts the A and B user identities andrecords the association in registry D.

F17 Step5. On recording the B and A user association, C determineswhether phone B is “ON/OFF net”, relaying the connection accordingly.This VOIP originated and redirected switching logic is depicted in theminimized flow chart that references FIG. 9 above. Similarly, the VOIPterminating logic described in FIGS. 14 and 15 are applied on Bconnecting to C and selecting associated user A.

In summary, by defining a “global proxy that virtualizes the legacytelephone network”, by hosting a new SIP domain, the Virtual Realm hooksthe disclosed methods and systems into any existing VOIP providerwithout change, where VOIP user A enters the telephone number B as a“Virtual VOIP user” and presses “call” to engage the new service.

However, whereas entering B as a conventional telephone contact(“14154125111”) in an unmodified service realm would result in a PSTNtrunked connection, one that terminates on the legacy network and incursbilling, addressing B on the Virtual Realm (“14154125111@dom.com”) asdisclosed, now hooks the new association and switching logic that alwaysterminates the call on the Internet, without change to the existingprovider and without charge to the caller.

Alternatively, a minimally modified service provider may automaticallyappend the Virtual Realm to a conventionally addressed phone callrequest and engage the system and method disclosed. In one suchembodiment an Outgoing Proxy Server (not shown) servicing the connectionrequest from VOIP user A to phone number B (“14154125111”) modifies therequested destination URI by automatically appending the DistributedVirtual Realm (“@1.1.1.5.2.1.4.5.1.4.1.dom.com”) to relay the connectionvia the delegated Inbound Proxy that delivers the disclosed methods andsystems as above.

FIG. 18

Continuing with the Virtual Service Realm described above, and inparticular expanding on the distributed DNS switching logic, FIG. 18further illustrates the Universal Edge caching data model.

F18 Panel 1: On receiving the connection request from VOIP user A tophone number B addressed on the Virtual Realm (“14154125111@dom.com”)the Inbound Proxy hosting the virtual domain rewrites the URI toincorporate the ENUM addressing model(“14154125111@1.1.1.5.2.1.4.5.1.4.1.dom.com”).

To the skilled artisan, it will be evident that the ENUM addressingmodel describes a highly distributed VPN (Virtual Private Network)architecture that switches telephone connections over a geographicallymapped E164 numbering plan, where switching nodes, in this instanceGateways and associated Registry Databases to the legacy PSTN, may bedelegated at any point along the reverse dotted path.

F18 Panel 2: In one such node delegation, all US telephone numbersaddressed on the Virtual Realm relays the connection to a “1 DOT” (US)zoned and delegated Inbound Proxy (Gateway C1 with associated RegistryD1). This distributed switching is achieved on configuring the wildcardDNS A record “*.1.dom.com” to map any phone number with the “1” countrycode prefix to the IP address of Gateway C1 servicing the continentalUSA.

Continuing with zone delegation in the continental USA, servicegranularity and distribution may be increased down to “state and city”following the NANP (North American Numbering Plan). By delegating zonesup to L6 (Level 6) in the dotted path, the following DNS routesconnections to a Gateway servicing San Francisco, Calif.:

L1 is TLD (Top Level Domain)=“con”

L2 is VPN (Virtual Private Network)=“dom.”

L3 is USA=“1.” and L456 is SFO CA=“*.5.1.4.”

Thus all phones numbers beginning with International Country Code “1”(USA) followed by area code “415” (SFO) are serviced on the DNS A recordfor domain “*.5.1.4.1.dom.com” pointing to the IP address of GatewayC5141 located in San Francisco, Calif. This zoning is graphicallydepicted in FIG. 33.

As described in FIG. 17 above, on receiving the inbound connectionrequest, C1 extracts the A and B user data in the connection requestheaders, recording the user association in the B client record. D1 thusautomatically records and assembles “Net contacts A” associated withtelephone number B, as shown.

F18 Panel 3: Similarly, all ZA telephone numbers addressed on theVirtual Realm relays the connection to a “72 DOT” (ZA) zoned anddelegated Inbound Proxy (Gateway C72 with associated Registry D27). Inthis instance distributed switching is achieved, by way of example, oninviting phone B “27824455566” on the virtual realm“6.6.5.5.5.4.4.2.8.7.2.dom.com” where the wildcard DNS A record“*.7.2.dom.com” is set to map any phone number with the “27” countrycode prefix to the IP address of a Gateway C27 servicing South Africa.

FIG. 19

The above embodiments disclose various selection methods made availableto telephone user B when connecting to automatically associated VOIPusers A via a generic gateway access number C. Describing now anembodiment that delivers tighter coupling between B and A, C dynamicallyassigns a unique regional and primary calling line identity to every A,indexed on the associated BA pair.

PICO

PICO (Peer Index Coupling) is a switching protocol that makes a routingdetermination based on both the dialed source and destination, which inthe context of this embodiment is calling phone number B and dialedgateway “Cn”, respectively. Comparatively, per the above disclosures,phone B to VOIP A connections made on dialing a generic gateway accessnumber C, switches by indexing contacts on the B number “source”,whereas conventional switching algorithms route on the dialed“destination”.

PICO is a dynamic pairing algorithm that anuses an extremely limitedshared number pool to uniquely slot assign a calling line identity toeach user A, when delivering “OFF net” notifications to phone B. FIG. 19highlights one such Paired OFF net ring back matrix.

FIG. 19 Panel 1: In this embodiment, on recording the associationbetween VOIP user A and phone B, gateway C records an additional datafield in the BA record, assigning a persistent telephone number aliasC(n).

As depicted, the first VOIP user A1 associated with B, has gatewayaccess number C1 assigned. The second user A2 has number C2 assigned,and so on. Each VOIP user A associated with phone B thus has a distinctnumber assigned, persistently recording and mapping the Internet useraddress A(n) with a gateway telephone alias C(n) in the B registry.

VOIP user A(1) presents telephone address C(1)

VOIP user A(2) presents address C(2) . . .

VOIP user A(n) presents C(n)

On determining “OFF net” status of phone user B as disclosed, gateway Cthen applies the assigned gateway number alias “Cn”, rather than ageneric access number “C”, to all notifications delivered as per FIG. 10above. When Telephone user B dials the C(n) alias, the gateway queriesthe B contact registry on the “B C(n)” dual index to uniquely resolvethe connection path back to Internet User A(n) without any additionalselection required by caller B.

FIG. 19 Panel 2: The PICO protocol results in a “sparse switchingmatrix” as depicted, where each VOIP user A associated with phone B,fills the next available slot (“sparse”, since many phone users B mayhave just a “handful” of contacts A).

PICO is predicated on the fact that a single telephone number “X” canrepresent “N distinct associations amongst N distinct AB pairs”. Thatis, A1 can uniquely present telephone number alias X to B1, while A2 canpresent the same telephone number X to B2, without identity conflict.Similarly A1 can present telephone number Y to B2 and still preserve theunique paired association.

The PICO protocol thus permits “a 1st or 2nd order” (10^1 or 10^2)number space, comprising just ten or one hundred numbers, to uniquelyaddress a user base “multiple orders of magnitude greater” (10^9). Thisis premised on Parabolic Social Geometry identified and depicted earlyon in this disclosure (see FIG. 38) that describes a “90:10” rulegoverning telecommunications:

“While Internet users number in the hundreds of millions and telephonesin the multiple billions, socially and typically 90 percent of usershave less than 10 distinct contacts on a regular basis”.

Consequently “10 base pairs”, 10 distinct shared regional telephonenumbers, can uniquely and locally pair a global Internet telephonycommunity, comprising hundreds of millions of users, with tens ofmillions of telephones in a “peer slotted indexed” fashion. Mapping VOIPusers A with associated telephones B, thus typically requires just 10numbers, mated in “BA pairs”.

FIG. 20

Continuing with the PICO protocol applied now to the “1:1” switchinglogic between Phone B and associated VOIP user A, FIG. 20 depicts thePaired OFF net ring back stepladder.

F20 Step10. VOIP user A(1) dials phone B(6).

F20 Step11. Connection is routed via optional outbound service Proxy(2)to PSTN gateway C(3) on what is a conventionally VOIP originated andPSTN terminated call.

F20 Step12. C receives the Internet outbound call originated from A totelephone B, and records the “BA user association” in registry D(4).This BA dataset includes the “B#” (the called party phone number) andthe “A@” (the Internet caller identity).

In addition to recording the B#A@ user association, registry D nowapplies the PICO protocol by programmatically assigning the nextavailable slot to user A in the “B(A)C” switching matrix as depicted inFIG. 19 above. Registry D thus assigns and returns the next phone numberalias “C(n)” available in the BA(n) array. If VOIP user A already existsin the BA record, D returns the number associated with the assignedslot.

F20 Step13. C establishes the forward connection towards B, sending anIAM (Initial Address Message) directed to destination Switch (5)servicing telephone B. The IAM header records PICO mapped number aliasC(n) as calling line ID.

F20 Step14. On receiving the IAM the switch forwards a call setupmessage to the B device, which in turn responds with an “alerting”message back to the switch.

F20 Step15. On receiving the “alerting” message, the switch returns anACM (Address Complete Message) back to gateway C, advising that the Bdevice is ringing. B now rings displaying calling line identity C(n).

F20 Step6. On receiving the ACM, C presents the ring back tone to callerA.

F20 Step17. Caller A hears phone B ringing.

F20 Step18. In one embodiment where B is determined to be “OFF net”, Cautomatically releases the connection to deliver a missed callnotification. In another embodiment on B failing to answer, C releasesthe CB leg and redirects the A to the Internet hosted messaging platformas per FIG. 13 above (not shown).

F20 Step19. On releasing the CB leg, B displays “1 missed call fromC(n)”

When B returns the missed call to C(n), the D record is then indexed onB# as the primary, and C(n) as the secondary, registry access key touniquely identify and retrieve user A(n). In one embodiment, B thusautomatically connects to A(n). In another embodiment, C first announcesthe Internet identity A(n) before connecting through, as per the“secondary caller line identification” feature disclosed above.

FIG. 21

The above embodiments describe dynamic VOIP user A and phone user Bassociation on the advanced identification of the latter by the formerand, in selected embodiments, reverse switching connections on therecorded associations when B connects to the cloud via Gateway C. Sinceany telephone may dial C, access to the disclosed methods and systemsrequire no legacy integration whatsoever.

While it is useful in delivering new service capability to legacysystems without change, the “personal mobile telephone number”, thedefinitive global digital user identity, lends itself to serviceextension only on direct legacy operator participation. Describing nowan alternate embodiment, VOIP users A register their existing (mobile)phone number as “net alias”, to deliver one such core legacy serviceextension.

One key omission in the IP Telephony architectural blueprint, was thelack of a globally defined dial plan that would permit legacy telephonenetworks and devices to dial and terminate connections “up into thecloud”, permitting IP telephony users to freely receive connectionsoriginated off net. By analogy, an IP dial plan would deliver “a staticnumerical equivalent to the IP domain name”, and it is this coreoversight that informs the technology described this disclosure.

Introducing a second global dial plan, would however result in “dualtelephony identity”, one for legacy and one for IP telephony devices, inand of itself introducing an undesirable “discontinuity”. Users wouldhave to be allocated a new phone number and distribute it amongst theirsocial network. Such a service would require administration by anumbering authority, which in all likelihood would attract unwelcomeservice fees. Nevertheless, an IP dial plan could be helpful inpermitting legacy devices to dial IP phones to call locally and connectglobally.

ALTO

ALTO (Authenticated Link To Owner) overcomes address duplicity andadministrative drag in allocating a new dial plan, by differentiatingand switching the existing known and ubiquitous directory “at source”,on a defining carrier service that directly integrates the methods andservices disclosed with legacy switching on a symbolic and universal“net prefix”. This new carrier enabled service galvanizes voice bylocking in terminating revenues “directly on the home operator network”,delivering a more fluid convergence business model, since interconnectis then via the PLMN rather than PSTN.

FIG. 21 depicts the abstracted core service elements in a symbolicprefix embodiment. Phone B (˜100) is any legacy telephone user. Symbolicprefix (#) routes conventional connections to the IP cloud (˜200) ratherthan to the legacy PLMN/PSTN cloud (˜300) which is the conventional plus(+) prefix route. A# is the phone number of VOIP user A, and while thismay be any legacy phone number, it is a mobile telephone in thedescribed embodiment.

F21 Step1. VOIP user A registers an existing personal phone numberA#which is stored in Registry D, together with the Internet useridentity A@.

Registry D thus records and associates phone number A with Internet useridentity A@. System and methods supporting phone number registration andauthentication are detailed in FIG. 26 through FIG. 29 below.

F21 Step2. Telephone user B dials phone number A prefixed in oneembodiment with the special “#” dial symbol. That is B dials “#A”.

In the illustrated embodiment, the gateway address prefix is thus asingle non numeric key, the pound (#) key, that then routes the call viathe Internet rather than the legacy network. In another embodiment thegateway address may be the “star” (*) key. In yet a further embodimentthe gateway address may be a double pound “##” or double star “**”prefix.

The network servicing user B has switching logic to route all suchsymbolically addressed connections to Gateway C. Such requisiteswitching logic, which may include setting “office wide triggers” in theOriginating Basic Call State Model (OBCSM) will be evident to theskilled artisan.

So for example,

+14154125111 a telephone number dialed in fully qualified e164international format, would route to the legacy telephony network anddevice, following the dotted gray path to the right

Whereas,

#14154125111 now routes to a VOIP associated counterpart whilemaintaining principle user telephony identity (the digits comprising thephone number), following the solid black path to the left.

That is once IP telephony users register their existing conventionaltelephone number with the service, they are locally contactable on thesame known phone number, prefixed as disclosed.

F21 Step3. Gateway C receives the IP routed connection and queriesRegistry D on called line identity A to determine the associated VOIPcontact A@.

F21 Step4. Gateway C resolves the user address A@ associated with thedialed destination #A, to the current IP endpoint where the VOIP clientis registered, ringing the destination with caller line identity set asB.

B thus dials #A to ring VOIP user A@.

Symbolically prefixing the VOIP client owner telephone number thus“virtualizes” (escapes) the global known E164 dial plan permittingprogrammatic and dynamic “1:1” (phone:VOIP client) switching, withoutassigning and dedicating a single new telephone number as Internetalias.

While the caller invokes the system and method described above bymanually prefixing a phone number, it is will be evident that the legacymobile network operator may, for example, on failing to locate a mobiledevice, automatically apply the address prefix as disclosed to switchand route a connection along the new IP path. Similarly, suitablymodified phone software may provide the user with the option to selectthe call routing, programmatically applying the universal address prefixwhen required.

In the automatic network rerouting case, either the mobile user or themobile network could selectively and conditionally forward calls to thenewly prefixed IP routed number, for example when the device is“roaming, unreachable, unavailable or busy etc.” where, in these andsimilar conditions, rather than redirecting callers to legacy voicemail,the connection would be transparently rerouted over IP, ringing theassociated VOIP client.

FIG. 22

Continuing with the ALTO switching protocol described above, withreference now to FIG. 22, which shows detailed logical interactionbetween prefixed core elements.

F22 Step1. VOIP user A(30) registers their existing phone number A#

F22 Step2. Registry D(40) creates an owner A#record, associating theInternet user identity with the registered phone number.

F22 Step3. Internet user A@ and A# are now aliased in registry D.

F22 Step4. Telephone user B dials #A and the connection is routed on the“#” prefix to internet Gateway C(20), establishing a first call leg BC.

On receiving the inbound connection from B, Gateway C may instantiate avirtual NANO client (B#) that serves as the VOIP client proxy to callerB.

F22 Step5. Gateway C queries Registry D on the dialed destination A.

F22 Step6. Registry D returns the previously associated VOIP useridentity A@ in the registered A# record.

F22 Step7. C invites and rings user A@ presenting B as the calling lineidentity, establishing a second call leg CA.

F22 Step8. C plays the ring back tone to caller B.

F22 Step9. A answers the call and C bridges the first and second legs,BC and CA, connecting phone B and VOIP user A in full duplex speech.

FIG. 23

Continuing with the ALTO system and method with reference to FIG. 23,which shows a Prefixed VOIP terminated call flow chart.

F23 Label ˜100: B dials #A.

F23 Label ˜200: The call is “#” switched and routed to PLMN gateway C.

F23 Label ˜300: First call leg “BC” is established.

F23 Label ˜400: C queries a service registry to determine if an A#recordexists.

Detailed phone registration is flow charted in FIG. 26 below. The dottedminimized flow chart at right highlights the registration basics:

F23 Label ˜410: As per the prefix related embodiment disclosed above,VOTP users A advance register their own phone number A# with theservice.

F23 Label ˜450: This phone number registration creates an A#record in aservice registry as shown.

F23 Label ˜400: [Yes]. IF A#record exists in registry, logic flows to˜500.

F23 Label ˜400: [No]. IF A#record does not exist, C notifies phone userA# of the service availability and notifies phone B that A is presentlyunavailable (process not shown).

In one such notification embodiment, C presents an engaged signal to Band sends mobile user A an SMS text message requesting their VOIPaddress by return, thereby wirelessly “forward propagating serviceregistration”. On receiving the text reply from A with their IPtelephony username, C stores the association in D for future reference.

F23 Label ˜500: C rings A@ with calling line identity set to Bestablishing second call leg CA.

F23 Label ˜600: C bridges the first and second call legs, BC and CA,connecting phone user B and VOIP user A.

FIG. 24

The ALTO “address aliasing” described above, permits conventionaltelephony users to dial IP telephony users on prefixing known phonenumber contacts.

FIG. 24 shows a Prefixed VOIP terminated switching stepladder.

F24 Step10. Phone B(1) dials #A

F24 Step11. Connection is routed on the #prefix to gateway C(3) via anoriginating Switch (2).

F24 Step12. C queries service registry (4) on the dialed destination A.Registry returns the VOIP user A@ associated with the previouslyregistered phone number A.

F24 Step13. C invites the associated VOIP user A, setting callingidentity to B. Connection is relayed to Inbound Proxy (5) which locatesthe IP address of user A. In one embodiment, Inbound Proxy determinationis based on DNS SRV records referenced on resolving the URI (thedestination VOIP user realm).

F24 Step14. During connection invitation, proxy (5) locates VOIP user A,assigns a media channel back to gateway C and rings destination clientA.

F24 Step15. On ringing A, system progress messages (“183 ringing”) flowback to gateway C, informing the gateway that the destination isringing.

F24 Step16. C presents ring back tone to B.

F24 Step17. B hears A ringing. Voice leg1 (BC) is established.

F24 Step18. When A answers, C receives call acceptance message (ok/200)and acknowledges.

F24 Step19. Second media circuit (leg2) is established between A and C.

At this stage, C may apply one of various bridging architectures, toconnect end users A and B, as described in FIG. 14 above.

FIG. 25

As described above, the symbolic telephone number prefixing system andmethod pivots the known dial plan on a new net “root prefix” that routesconnections to the authenticated IP owner. FIG. 25 illustrates thedisclosed E164 dial plan virtualization schema.

Legacy telephone (˜100) dialing E164 plus (+) prefixed phone numbers(˜200) route along the horizontal axis to the legacy and geographicallymapped telephony networks (˜250). The disclosed method and system showslegacy telephone dialing the IP root (#) prefix (˜300) that routesconnections along the vertical axis into the IP cloud (˜350) via mediagateway (not shown) and registry database mapping the dialed numbersinto associated IP addresses.

Legacy E164 geographically maps the global dial plan according to theITU specified numbering system, where leading dialed digits map tocontinents and countries. By example, and as illustrated, common countryleading (and secondary) digits include 1=US, 2(7)=ZA, 3(3)=FR, 4(4)=UK,5(5)=BR and so on.

The Plus (+) prefix represents the international outbound dialing codeplaceholder, utilized by GSM and other mobile telephony systems,permitting users to store phone numbers in a universal format whileroaming between countries. The visited network then substitutes theinternational outbound dialing code for the host country.

For example, Plus (+) in the US would be replaced with the international“011” dialing code, whereas in ZA “09” would be substituted. Plus (+)thus symbolically unifies the international outbound dialing codesamongst different regions, without which mobile users would have to dialdifferent prefixes in each country visited.

Virtual E164 permits users to symbolically address an Internet MediaSubsystem and Gateway by prefixing the conventional telephone number, inone embodiment, with a Pound (#) rather than a Plus (+) sign, andthereby switch connections up to the IP cloud rather than over thelegacy network which incurs international tolls.

While the system and method disclosed in this prefixing embodimentreference a privately hosted Registry that records the dialed A# and theauthenticated A@address mapping, it will be evident that the legacynetwork operator may utilize existing public or private registryservices, in particular resolvers based on ENUM (IETF RFC 3761), whichmap specially formatted phone number domains into associated Internetcommunication services.

In one such embodiment, the IP media gateway receiving the “#” switchedconnection may perform a DNS query on the ENUM assembled domain, whichreturns NAPTR resource (URI) records. If multiple NAPTR records arereturned, the gateway uses record field values, including “Order,Preference and Services”, to select the most appropriate destinationURI.

For example:

(i) Legacy Phone B dials #14154125111 which routes to IP Gateway C

(ii) C assembles the ENUM domain “1.1.1.5.2.1.4.5.1.4.1.e164.arpa”

(iii) C queries a DNS server on the ENUM domain

(iv) DNS returns NAPTR records describing client supported services

(v) C selects the appropriate URI and routes the connection accordingly

It will be evident to the skilled artisan that such an ENUM resolvedconnection supports switching multiple services, bearers and protocols.For example, by similarly addressing text and multimedia messagingservices (on the #prefixed phone number), users and carriers can switchand route a multitude of services to the associated IP client ratherthan the legacy device.

Further, in the absence of the appropriate NAPTR record and URI toservice the requested connection, switching may revert to the legacyroute as the fallback path. In “virtualizing” the dial plan thus, thedisclosed symbolic addressing permits legacy telephone users tocommunicate globally over the IP Next Generation Network, all on a wellknown user identity at local cost.

FIG. 26

The ALTO method and system disclosed is predicated on an IP userregistering their existing telephone number, referred to in the currentembodiment as “A#” as “net alias”. As such, the system requires a secureauthorization mechanism to ensure that the registering party owns thenominated telephone number A#. The authorizing methods and systemsfollowing utilize an “out of band” challenge response mechanism. Onesuch system is illustrated in FIG. 26 (F26), which shows a RandomizedTelephone number authentication flow chart.

F26 Label ˜100: VOIP user client A@begins a process to authenticate andbind telephone number A# as alias. User A enters telephone number A#either directly via the registered VOIP client software or via anassociated website.

F26 Label ˜200: On receiving the nominated phone number, the VOIP systemgenerates an “out of IP band” challenge, transmitted to telephone deviceA# over the legacy telephony network. In the illustrated embodiment,this challenge is presented via a missed call.

F26 Label ˜210: The missed call is a challenge sent directly to thescreen of the A#telephony device. The authenticating service generates arandom caller identity, depicted by “555xxxx” (IDx), where either theentire number or the last “x digits” comprising the telephone number,are randomly generated.

F26 Label ˜220: The system deposits this randomly generated identity onthe telephone device (A#) by setting Calling Line Identity as IDx,automatically ringing and then substantially simultaneouslydisconnecting a call addressed to telephone A#, in a flow charted methodsimilar to that illustrated in FIG. 20.

F26 Label ˜230: The system records this randomly generated telephonenumber IDx, together with the nominated phone number A# and a validitytimer in a registry database 240 in a registry 2600 for futurereference.

F26 Label ˜240: The registry database 240 thus reflects an A#record,together with the random IDx phone number challenge and an expiry date.The expiry date is typically computed by “adding the timer interval tothe current date and time”, as at the completion of the challengeprocess.

F26 Label ˜300: On depositing the missed call challenge, the systemrequests user A to respond by entering the IDx calling identity,transmitted over the legacy telephony network and displayed on thetelephone device, back into the PC.

Since the challenge is carried over a network and channel that isdistinct from the PC requesting the response, the mechanism issafeguarded from interception by an eavesdropping third party.

Further, given that the challenge is typically presented on a devicethat is independent of the VOIP system, the user is required to havephysical access to the telephony device, B, in order to retrieve thechallenge, and successfully respond via the PC screen requesting the IDxcode.

In a timer based authentication method, the “validity timer” associatedwith the challenge response requires the user to respond to the IDx codechallenge within a predetermined interval.

Typically, the validity timer is set to expire in a short interval, forexample, less than 60 seconds after the user enters the nominated phonenumber, since the challenge generally presents on the phone display mereseconds thereafter.

Establishing ownership is thus predicated on the alleged telephone owner(User A) having the nominated phone (B) “in hand”, since “near realtime” response is required in order to respond to the challenge.

F26 Label ˜310: The authenticating service checks the registry database240 to determine whether the response entered matches the challengesent, in the time allotted. The timer validity may be determined byensuring that the current time is earlier than the expiry time, ascomputed in ˜240.

F26 Label ˜400: [No]. IF the response does not match the challenge orthe timer has expired, the authorization fails as depicted by circle “X”410.

F26 Label ˜400: [Yes]. IF the response entered matches the challengesent in the allotted time, the authorization succeeds and logic flows to˜500.

F26 Label ˜500: Since the response and challenge match, it may bedetermined with a high degree of certainty, that IP User A “owns”, or atminimum, has direct access to telephone A#.

On thus successfully authenticating the nominated phone number, thesystem records the association between telephone number A# and IP useridentity A@ in a service registry 2600. This results in an A#phonerecord with associated A@IP user identity. Note that in the currentnon-limiting example, the phone number of User A (A#) is the same as thephone number (B#) of User A's phone (B).

F26 Label ˜600: VOIP User A now has telephone number A#set as alias.This authenticated phone number is used to switch and route connectionsoriginated on legacy telephones (B's), to the associated VOIP client(A@'s), by symbolically prefixing the directory telephone number asdisclosed in the above described ALTO method and system.

FIG. 27

FIG. 27 (F27) depicts Randomized Telephone number entity relationships,presenting the above authentication process from the perspective of theVOIP user registering the phone number.

F27 Panel 1: VOIP user client A, either directly via the VOIP clientsoftware or via an associated website, is requested to enter 2700 phonenumber B# to authorize. By example, VOIP user “AK@sip.net” enterstelephone number “14154125111”, via the Internet connected PC screen2702A as shown (expanded).

F27 Panel 2: On receiving B#, the system generates a random telephonenumber challenge B(x), as depicted by “555xxxx” and described in FIG. 26above in regards to IDx. This user entered telephone number B# togetherwith system generated challenge B(x) is stored as a “B authenticationrecord” in an authentication (AUC) center (2600 of FIG. 26) firsubsequent retrieval and reference.

The AUC may be a physically separate storage unit coupled to theauthenticating node. Alternatively the AUC may be a local store or amemory based store on the servicing Internet authentication node itself.Optionally, the system allocates a “validity” timer, the time untilwhich this challenge remains valid.

F27 Panel 3: The system deposits a “B(x) missed call” on telephonedevice B. That is via an associated legacy Gateway C, the systemsprogrammatically rings telephone b with caller line identity set tob(x), and disconnects the call momentarily thereafter, preventing B fromanswering. Phone B displays “1 missed call from 555xxxx” (where “555xxxxis an example of IDx B(x), as shown.

This missed call thus presents a random “out of band” challenge directlyto the display screen of telephone B. The telephone thus requires nospecial software, other than the standard capability to present calleridentity, in order to successfully challenge the owner.

F27 Panel 4: On challenging the user, the Internet authenticatingservice presents a follow on PC screen requesting User A to enter themissed call digits as response 2704. As illustrated, User A enters thesame “555xxxx” code received on phone B back into the PC screen 2702B.

F27 Panel 5: On the receiving the 2704 code via the PC, the VOIP systemcompares it with the B(x) code stored in the “B authentication record”as shown in Panel 2. If the B(x) code sent via the missed call eventmatches the 2704 code entered at the PC within the time allotted,authorization successfully completes (see F26, step 500).

On successfully matching the challenge and the response as above, thesystem records the now authorized telephone number and the associatedVOIP user identity in a service registry, as described above withreference to F26 step 600. As illustrated, phone number “14154125111” isnow the numeric dialed alias to VOIP user “AK@sip.net”.

FIG. 28

In an alternate authentication embodiment, the missed call challenge(the “ping”) presented to the registering telephone, may be returned(the “pong”) via the legacy network back to the authenticating servicewithout requiring manual code reentry at the PC. One such “ping pong”system is depicted in FIG. 28 (F28), which shows a Returned Telephonenumber authentication flow chart.

F28 Label ˜100: Registered VOIP user client A@enters telephone numberB#into an authentication service, via an Internet connected PC.

F28 Label ˜200: On receiving phone number B#, the PC programmaticallyconnects to the “ping pong” service.

The following “210˜260” logic, in the dotted frame, executes in parallelwith the main authorization loop “300˜400”.

F28 Label ˜210: <ping>. The service deposits a missed call challengeonto device B as described with respect to F26. User A is then requestedto return the missed call received on telephone B.

In one embodiment, the missed call identity is randomly generated on agateway prefix, to deliver “direct in dial extensions”, where the last“x digits” of the gateway address are randomly generated. In anotherembodiment the missed called identity is static, representing a completegeneric gateway address.

F28 Label ˜220: <timer>. The system initializes a countdown timer T inseconds. This timer automatically counts down to zero, clocking thereturn call from the registering phone number B#.

F28 Label ˜230: <monitor>. The system monitors the gateway identified bythe challenge, awaiting the return call from B.

F28 Label ˜320: [No]: IF the timer has counted down to zero (expired)then authorization fails, as depicted by circle “x” 330.

F28 Label ˜320: [Yes]: IF the timer is still positive (ticking)authorization continues at ˜400.

F28 Label ˜300: The authentication service enters a programmatic loopthat polls 400 the gateway for the return call within the specifiedtime.

F28 Label ˜240: <pong>. Telephone B returns the missed call which routsto the servicing gateway C.

F28 Label ˜250: <detect>. The monitoring gateway C detects the incomingcall from phone B.

F28 Label ˜260: <record>. The monitoring gateway C records the returncall from phone B in a service registry (not shown).

F28 Label ˜400: The authenticating service checks the authenticationregistry (not shown) to determine whether the servicing gateway hasdetected and recorded a return call from B.

F28 Label ˜400: [No]. IF no such return call was detected and recorded,logic loops back to ˜320

F28 Label ˜˜400: [Yes]. IF phone B did return the missed call logicflows to ˜500.

F28 Label ˜500: Since B returned the missed call challenge within theallotted time, it may be determined with a sufficiently high degree ofcertainty, that phone B belongs to user A.

On successful authentication, the system registers the associationbetween telephone number B# and IP user identity A@(registry not shown).The “ping pong” system and method, delivers frictionless “2 clicks”authentication, without any manual code entry required, on returning acall.

FIG. 29

Highlighting the “ping pong” authentication method and system in a moregraphic presentation, FIG. 29 (F29) illustrates the returned telephonenumber entity relationships.

F29 Step1. VOIP user client A@, either directly via the VOIP clientsoftware or via an associated website, is requested to enter 2900 thephone number B#being authorized. By example, VOIP user “AK@sip.net”enters telephone number “14154125111”, via the Internet connected PCscreen 2902A at top left, as shown (expanded).

F29 Step2. The system deposits a “missed call” on telephone device b viaan associated legacy Gateway (c) as described with respect to F28 above.In the illustrated embodiment, the missed call line identity is set to astatic gateway telephone number “7132421581”, as displayed on the legacyB device screen (expanded).

On depositing the missed call, the system records the expected returncall telephone number B# in a registry (such as described in referenceto F28, registry database 240 in a registry 2600, not shown). Typically,this register is memory resident at the gateway C. The gateway Cmonitors incoming calls to determine if one originates from telephone B.

F29 Step3. The system starts a countdown timer T(−) by example, a “30second” timer as depicted in the anticlockwise semi circle in PC screen2902B top right. This timer automatically counts down to zero at whichpoint the authentication process expires. The countdown timer expireswhen the countdown timer has counted down all of the given time withwhich the countdown timer was initialized. While the timer is active,the system polls the gateway for a returned call from B.

F29 Step4. User A is requested to return the missed call on phone B.Missed calls are typically returned by pressing “Send” (Connect) on amobile telephone. On touchscreen telephones, the user may touch themissed call displayed to return a missed call, whereas on fixed linetelephones, the user may press a dedicated “return call” key. The returncall is routed back to the gateway C identified by the return dialeddigits.

F29 Step5. On the receiving the returned missed call, the gateway Cmatches the B calling line identity B# in the registry of expectedreturn calls and checks that the timer is still valid (non zero). If thecalling identity B#matches a registry entry and the timer is valid,authorization successfully completes and the registry entry is cleared.Telephone B is then authorized, depicted by the ticked (checked) phonenumber in the PC screen 2902B expanded at top right.

F29 Step6. On successfully matching the return call within the allottedtime, the system records the now authorized telephone number and theassociated VOIP user identity in a service registry, such as registry2600, as illustrated. By example, phone number “14154125111” (A#) isregistered as the numeric dialed alias to VOIP user “AK@sip.net” (A@).

FIG. 30

The ALTO methods and systems described are contingent on legacy carrierparticipation in allocating the symbolic prefix required to switch androute connections over IP. Alternatively, an Independent ServiceProvider may offer equivalent IP switching and routing functionality onan unassigned “country code prefix”, as per the ITU E164 dial planallocation.

Such a virtual and carrier independent IP switching prefix, permitsusers to freely bind their existing IP services to an authorized phonenumber without the legacy network operator having to dedicate, and thus“consume”, a Top Level Digit in the cellular keypad matrix to performthis function (for example, the “#” key). FIG. 30 illustrates thisGlobal IP phone number assignment in one embodiment.

UNNO

In the illustrated embodiment, a Universal Net Number (UNNO) isformulated by prefixing an existing authorized telephone number with anewly assigned ITU country code prefix, to deliver an IP switchedabstraction to telecoms. In one such embodiment, this virtual operatoris defined on an unassigned ITU “triple digit country prefix” (threeidentical digits) also known as an “easily recognizable number” (ERN).One such ERN is the presently unallocated “+777” country code, which inthe disclosure would now define a Virtual and Global IP Operator.

In one embodiment, the ERN may be dialed as a fully qualified E164address (including the International “+” prefix) that would route onGlobal Title Translation (GTT) to an assigned Internet Media Subsystem(IMS) gateway. It would be clear to the suitably skilled switchingartisan, that on programming the relevant routing table entry into theswitching subsystem, the originating operator would thus route all such(+777) prefixed connections to the requisite IP Gateway, rather thanconventionally switching the connection internationally over legacynetworks.

Alternatively, the ERN may be dialed as a “locally formatted number”(without the “+” prefix) that the operator would then normalize to thefully qualified E164 format. In either instance, any “777” prefixedtelephone number would return a routing destination equivalent to an SS7point code identifying the IMS node on the originating carrier network,switching the connection thus over IP to the associated VOIP user, asdescribed below.

The UNNO allocation and IP service association, continues and extendsthe ALTO authorization process described above. In the illustratedembodiment, the service is conducted by the UNNO numbering authority“www.dom.com”, accessed via an Internet connected PC (as shown). Whereasthe authorization processes described previously automatically linkedthe registering VOIP user, UNNO permits any user to associate any IPservice to the authorized number.

F30 Step1. On successfully completing a telephone number authorization,the user is presented with the service association screen (upper panel).This screen permits the user to associate a plurality of VOIP user namesand services with the UNNO allocated and authorized number. Theillustration shows authorized phone number (“14154125111”) being linkedto two user supplied VOIP addresses (“AK@SIP.NET” and “AK@VOI.COM”) viathe “+777” Virtual IP Operator Prefix.

F30 Step2. On entering the services to be linked, the system assemblesand records ENUM NAPTR resource records (abstracted as shown) on theauthorized telephone number domain. In the illustrated embodiment, theserecords are hosted on a private domain (“dom.com”) however they maysimilarly be hosted on the ENUM public domain (“e164.arpa”). Thislinkage, together with the associated DNS ENUM resolution would, in oneembodiment, result in both VOIP “extensions” (AK@SIP.NET andAK@VOIP.COM) ringing on dialing “+777 14154125111” from any legacytelephone.

On switching such a “777” prefixed connection, the servicing Gatewaywould perform a DNS ENUM resolution on the subscriber digits received inthe connection setup request (14154125111) to retrieve the NAPTRassociated resource records, and connect to the appropriate IP service(as per the description pertaining to FIG. 25 above).

It will be appreciated that a simple software modification to mobiletelephones can permit “button press” IP call switching, by automaticallyprefixing conventional telephone numbers with the common carrier andindependent global prefix disclosed.

In one such software modification, on selecting a dedicated key orsoftware feature, any standard telephone contact address may beautomatically and transiently modified to incorporate the GlobalOperator IP prefix prior to issuing a call setup request (that is priorto signaling “Connect”).

For example, selecting a conventionally stored contact “AK” (telephonenumber “+14154125111”) and pressing and holding a suitably modified key(for example, the “#” key), would programmatically transform the contactphone number into the symbolic ALTO (#14154125111) or UNNO(+77714154125111) prefixed format and transmit the connection request ina single step.

FIG. 31

In an alternate IP switching prefix system and method that forgoestelephone registration and authorization, two legacy telephones may beautomatically paired over the Internet through a discovery process thatdynamically establishes NANO numbered clients and exchanges contactdata. FIG. 31 depicts an Example Telephone pairing over IP.

In the illustrated embodiment, IP switching is on the symbolic “#”prefix to the dialed destination. In another embodiment the prefix isthe virtual IP operator “777” ERN as disclosed earlier. In eitherinstance, no advance telephone number registration and authorization isrequired.

F31 Step1. Telephone user A(10) dials “#B. Originating switch (notshown) routes the connection on the symbolic prefix to Gateway C(20),establishing a first communications leg AC.

F31 Step2. C instantiates a NANO client (A#) that serves as the VOIPclient proxy to caller A. NANO client proxy instantiation andregistration is described in detail above, with reference to FIG. 2.

F31 Step3. C registers proxy A with service Registry D(40), whichrecords the NANO client IP address and Port (not shown). On registrationC queries the registry on the AB connection setup data.

F31 Step4. D determines that A and B client records do not exist. Dautomatically creates A and B client records, exchanging contactsbetween them. That is on receiving a connection request from phone A tophone B, registry D creates numbered NANO account A with B as contact,and conversely, creates NANO account B with A as contact.

Since telephone B is currently “OFF net” as per the discloseddefinition, that is B has not connected to a service gateway andconsequently has no Proxy IP address and port registered, A is put onhold. Holding is typically indicated to A by playing music or playing aspecial tone.

In the illustrated embodiment A and B have rendezvoused the connectionusing collateral communication channels such as email, text and instantmessaging. Alternatively, C may automatically notify B that A is“holding the line”, prompting B to establish connection by calling “#A”.

F31 Step5. Telephone user B(30) dials “#A”, and is similarly routed toGateway G(50), establishing a second communications leg BG.

F31 Step6. G instantiates a virtual NANO B client, that serves as VOIPclient proxy to caller B.

F31 Step7. G registers proxy B with service Registry D(40), whichrecords the NANO client IP address and Port (not shown). Onregistration, D determines that A and B client records exist and that Ais registered and “ON net”.

F31 Step5. G invites Proxy A. Since A explicitly dialed B, theconnection is automatically established and announced. Caller B hears ashort ring back tone played by G. Caller A hears a special connect tone,played by C. Connection establishment bridges the first and secondcommunication legs, AC and BG.

F31 Step9. Telephone users A and B are paired and connected.

FIG. 32

The advanced association between VOIP users and legacy telephonecontacts has focused on service delivery within the realm of a singleservice provider. In these examples, the registry recording the IP andTelephone association has thus far been presented within the context ofthe service provider network.

The discoverable nature of identifying Internet users, by reversing acalling line identity into their online address books to determineassociation, broadens the application of the disclosed methods andsystems. Alternatively, the gateway servicing the telephone connectionmay query a plurality of registries (directories), associated with aplurality of Internet service providers.

These online registries may collectively be referred to as the “Websocial directory” and the amalgamation of users associated with aparticular telephone number across these multiple service domains, maybe termed a “mash up”. Querying the web social directory to discover anddynamically assemble a mash up of Internet users that have recorded aparticular telephone number as a contact, is illustrated in FIG. 32,which depicts an Example web social telephony mash up.

F32 Step1. Telephone user B(10) dials Gateway C(20) to establish a firstcommunications leg BC.

F32 Step2. C queries online directories D(40) from multiple serviceproviders (SP), using the B calling line identity as access key toidentify VOIP users A(30) that have the B phone number listed.

F32 Step3. In the illustrated example, D returns users A(xyz) who havephone number B listed as contact Users A(xyz) may differ in the VOIPprotocol adopted by their respective service provider.

F32 Step4. C plays a contact selection menu to B as depicted by the boldrectangular speech bubble. Various menu presentations and user selectionmethods are described throughout this disclosure.

F32 Step5. B selects user “A” from the A(xyz) list of users presented.

F32 Step6. C rings A with tailing line identity set to phone number B,establishing a second communications leg CA. While the illustratedembodiment rings A using the open standard “SIP Invite” method, it willbe evident that C would support the various protocols required in orderto contact users associated on different technology platforms.

F32 Step7. A answers the call.

F32 Step8. C bridges the first and second communication legs, BC and CA.

F32 Step9. A and B are thus connected.

FIG. 33

Alternatively, association may be delivered between VOIP users andlegacy telephone contacts may be utilized to establish an anonymous“instant callback channel” on the net, mapped directly to a selectedtelephone number.

In this alternate embodiment, VOIP user A is not required to sign intoservice with any particular provider and username. Rather, on entering adesired contact phone number, VOIP client software defines a logical “Bnet extension”, a new window or tab, that “blind registers” (withoutuser authentication) the current IP address of the anonymous client on aVirtual Realm.

The Virtual Realm is distributed over a B telephone ENUM domain, asdescribed in FIG. 17 above, that transports, registers and caches the Bnet extension directly at an edge Gateway servicing the call backtelephone user. FIG. 33 illustrates establishing one such Instantcallback service channel.

SOLO

Conventionally, a VOIP Gateway with DID (direct in dialing) capability,routes an incoming call, from the legacy network to an IP ServiceProvider, on the dialed destination. Call routing in the conventionalcase is thus determined by mapping the DID dialed number to a registeredURI. Typically, this URI resolves to a location server that then locatesthe VOIP user associated with the DID number.

The anonymous call back embodiment introduces a unique new switchingalgorithm that dynamically assembles the URI at the DID Gateway on“source” (B) rather than dialed destination (C), constructing theidentical ENUM URI string that the anonymous client registered under.This dynamic URI construction routes and distributes the incoming calldirectly to the VOIP registry containing the location of the anonymousclient.

“Switched On Line Origination” (SOLO) thus programmatically assemblesthe inbound call path and URI “rcfcrcncing self”, the B callingidentity, dynamically routing and resolving service on the resultantENUM realm without any additional switching logic required. By example,if telephone number B is “14154125111”, the anonymous extensionregistration and callback route is established on URI:

14154125111@1.1.1.5.2.1.4.5.1.4.1.dom.com

F33 Step1. Anonymous VOIP user A(30) enters phone number B(“14154125111” as illustrated).

F33 Step2. VOIP A client software opens a B titled and thus linkedwindow, the “B net extension”, registered over the ENUM Realm“14154125111@1.1.1.5.2.1.4.5.1.4.1@dom.com”.

In one embodiment, on registering the extension the client software setsan “automatic answer timer” (T) for the selected extension. If a returncall from B is received within the timer period, the call isautomatically answered. This automatic return call pick up within thespecified time, ensures that the most recently established extension isprimary.

For example, user A and B negotiate a connection utilizing a collateralcommunication channel, such as SMS. At the scheduled time, A opens theanonymous extension and the automatic answer timer is set to 90 seconds.This extension thus establishes itself as “priority answer” over anyearlier extensions, whose timers would typically have already expired,since B has rendezvoused most recently with A.

If B is on an active call with an earlier established extension, B maycall the newly established extension and selectively conference the twoextensions using “multi call” control features supported on the legacynetwork. If no active automatic answer is active, the extension thatmanually answers establishes connection. In selected embodimentssupportive client software may permit A to reactivate the automatictimer in order to “prime” the extension for pick up.

F33 Step2 x. As illustrated and by example in expanded view (dottedellipse), the ENUM Realm automatically resolves to an edge Gatewayservicing San Francisco (SFO), California USA. This Gateway is zoned anddelegated to Level 6 (L6) in the reverse B dotted path, which describesthe home base for all “1415” prefixed telephone numbers. Granular zoningon the NANP is detailed in the description related to FIG. 18.

F33 Step3. The ENUM registration resolves via DNS to the delegatedRegistry D(40) that records the IP address location (@P) for eachanonymous VOIP user in a B number record (the “B net extensions”).Optionally, each extension registration is time stamped T(n).

Edge Registry D(40) and Gateway C(20) are typically one and the samenetwork node and are herein referenced interchangeably.

F33 Step4. Telephone user B(10) dials Gateway C(20), establishing afirst communication BC.

F33 Step5. C assembles the B ENUM URI as above and rings the resultantnet extension to establish a second communication CA. In one embodimentC group rings all B registered extensions in parallel (“group” asshown). In another embodiment C rings the last extension to register,time stamped T(n).

On C ringing the extension(s), the associated A client window presentsthe incoming call.

F33 Step6. The A(n) extension answers, as illustrated. This call isanswered either automatically, if the automatic answer timer is stillactive, or manually on the user accepting the call.

F33 Step7. C automatically disconnects any additional extensionsringing.

F33 Step8. C bridges the first and second communication, BC and CA.

F33 Step9. A and B are thus connected.

FRISB

One service that serves to highlight the industrial applicability of thedisclosed systems and methods is “FRISB™ dot corn,” which is a VOIPtechnology implementation and service that uniquely applies game theoryto telecoms.

Modeled on the popular analog game “Frisbee”, digital FRISB players“spin virtual discs around the world, from net to phone, catchingreturns to speak free”. In reversing connections thus back into thecloud, FRISB practically reinvents the Internet Phone. FRISB inparticular overcomes the 3 main inhibitors to mass service adoption:

1. Downloading

2. Registering

3. And Paying

This direct to consumer service, hosted in the cloud and accessedthrough a standard web browser, applies the core methods and systemsdisclosed. By embedding a SIP client directly into a web page to voiceenable a browser, FRISB allows users to direct their browser to“www.frisb.com” and instantly gain remote access to global communicationwithout downloading, without registering, and most significantly,without paying.

The embedded web phone applet may be developed using software platformsand environments including amongst others, Sun Microsystems Java™, AdobeFlash and Air™, Microsoft ActiveX and Silverlight™ and the emerging“HTML5”. Developments may further utilize HTTP “tunneling”, technologythat encapsulates VOIP telephony commands and media streams through thestandard web browser “port 80” and related communication protocols.

Dubbed “Voice on the fly”, FRISB delivers “the highest abstraction oftelephony on the net”. Whereas search engines permit users to freelylocate almost any information stored on the web, by typing a few words(the search) into a browser, FRISB permits any user to freely connect toany legacy telephone in the world, by typing a few digits (the phonenumber) into their browser.

Fundamentally, while the service is provided free to net players, ituniquely generates revenue on the “interconnect” generated by the fixedand mobile phone return calling via a local legacy service gateway thatinterconnects VOIP. The unique FRISB business model is thus “freeforward, pay back”, as in “free contact on the spinning the discforward,” and “payment on the phone user calling back”.

Modeled on seamless callback functionality where the phone user pays fora local call and is instantly connected to a global community who talksfree, FRISB enables the VOIP industry to “leap over the penny gap”, thegap in service that demands payment for calling a legacy telephone andestablishing an “off net” conversation.

FRISB and the underlying methods and systems disclosed, thus uniquelyconverges “net social” (free) with “cell capital” (paying) to deliver adefinitive “fixed mobile convergence” service that reverse connectsconventional telephones to voice enabled browsers, without changing themass user behavior and core service propositions in either realm.

FIG. 39

Describing the “game” in one embodiment with reference to FIG. 39, FRISBis played between an Internet web browser and a conventional telephone.

F39 Step1. Web user A directs the browser to “frisb dot com” and entersa conventional telephone number B# (“14154125111” as illustrated),sending the disc to the contact they wish to establish communicationwith.

The web home page screen presents a colored disc progression along thelower edge, indicating the various states through which the gameprogresses. These states are described in FIG. 40 below.

F39 Step2. On entering the B player phone number, the web service “popsup” a second browser window titled B (“14154125111”) that references anddownloads the embedded SIP phone applet.

The applet is automatically configured to register the B “discextension” as per the SOLO method and system described above. Oncompleting automatic service registration, FRISB has established alogical callback path from telephone B to the web browser window withsame title.

Telephone B receives a “virtual disc”, presented as a callbacknotification and prompt. In one embodiment this call back notificationis a “missed call” with the local service Gateway that the discregistered into set as the caller line identity.

FRISB thus automatically registers and logically creates a “direct voicechannel back to the browser embedded SIP endpoint”, all on the userassociating the Internet browser with the address of conventionaltelephone subscriber. When the associated telephone calls a localservice gateway, the browser rings and the telephonic connection isestablished in reverse.

FIG. 40

FRISB uniquely presents a universal interface using the elements oflight (color) and sound to signal call state and progression. The discalters state by changing color and flying (animating) into view.Utilizing the universal signaling color spectrum, the disc progresses instate from Red to Orange to Green and finally to Blue. Describing thisprogression in more detail with reference to FIG. 40 (F40):

F40 Screen1: on successfully embedding and initializing the applet, thewindow displays a “disc” colored RED.

Red indicates the state of “wait”.

F40 Screen2: on loading the applet the service automatically registersthe browser embedded VOIP endpoint A with the media server that is usedto route the call back to the browser address captured in registration.On successfully registering the browser endpoint for callback, the Btelephone is pinged (as described below) and disc goes Orange.

Orange indicates the state of “shift”.

F40 Screen3: when the phone number B previously identified to theservice by user A calls the access gateway, the connection is routedback to the browser which then rings. On successfully ringing thebrowser the disc goes Green.

Green indicates the state of“go”.

F40 Screen4: if the browser rings back within a short “delta” timeframe(for example, within 90 seconds) from user A spinning the disc, theembedded phone automatically answers the return call, otherwise theinternet user clicks the now green disc to pickup the call. Onsuccessfully answering the call the disc goes Blue.

Blue indicates the state of “communication”.

FRISB is now “flying” and full duplex audio circuits are establishedbetween the browser and the phone permitting both players to talk. Sinceusers typically rendezvous the connection establishment, the callback isreturned within the automatic answer timeframe and the call isautomatically completed. A clock showing current call time in “hoursminutes and seconds” is displayed in the window title (00:00:01 asshown).

Disc signaling and associated “color spectrum” thus transitions from:

Red (service and dial tone establishment) to

Orange (dialing and initiating callback) to

Green (ring back) and finally to

Blue (in communication)

This all without the sender A having to interact with the disc elements.This timely and automatic progression in connection state gives theimpression that the call is actually being established in the forward(AB) rather than the reverse (BA) direction, since the “ring back” fromB typically presents within an acceptably short time frame to beinterpreted as a forward established “ring back tone” (the B party isringing) rather than what it in fact is, a “ring back” (B calling backand ringing A).

When either party disconnects the call, the disc returns to Orange.

If the return call from telephone player B arrives after the automaticanswer period, that is it arrives later than what would be interpretedas a forward established connection from A to B as described, the discpresents as a “ringing phone”, repeatedly “flying into view” (animating)while ringing, awaiting the user to click it, to “catch it” and answerthe call.

If the user answers the call, for example by clicking the now Greendisc, the connection is established and the disc changes color to Blue.If the call goes unanswered after a predetermined period of time, thedisc returns to Orange and indicates a “missed call”. If user A wishesto “spin the disc again” and thereby ping the caller B for callback onceagain, clicking the now Orange disc restarts the game and reengagesautomatic pickup on the expectant return.

FIG. 41

In particular with reference to the SOLO embodiment, the FRISB phoneclient registers on an ENUM Realm that dynamically and geographicallydistributes the game over the E164 dial plan. FIG. 41 illustrates threesuch distributed registrations and callback paths:

F41 Disc A1. URL www.frisb.com/14154125111 registers disc to:

14154125111@1.1.1.5.2.1.4.5.1.4.1.frisb.com

The above URL resolves via DNS to the delegated Registry D1 servicinggateway C1, locally registering all USA disc destinations (E164 leadingdigit “1” equals United States of America).

F41 Disc A2. URL www.frisb.com/27832226287 registers disc to:

27832226287@7.8.2.62.2.2.3.8.7.2.frisb.com

The above URL resolves via DNS to the delegated Registry D2(7) servicinggateway C2(7), registering all ZA disc destinations (E164 leading digits“27” equals South Africa).

F41 Disc A2. URL www.frisb.com/972523564110 registers disc to:

972523564110@0.1.1.4.6.5.3.2.5.2.7.9.frisb.com

The above URL resolves via DNS to the delegated Registry D9(72)servicing gateway C9(72) locally registering all IL disc destinations(E164 leading digits “972” equals Israel).

On disc registration the servicing Gateway deposits a missed call on thetelephone described in the registration headers. This missed call isprogrammatically delivered as described above.

On receiving the return call from the registered telephone the Gatewayassembles the SOLO URI, formulated on the B calling line identity asdisclosed, that routes the connection back to the web browser. TheOutbound Proxy for the FRISB realm (Registry in the illustratedembodiment) resolves the URI to the IP address of the browser thatregistered the disc.

FIG. 42

To the skilled VOIP artisan, the elastic nature of the VPN (VirtualPrivate Network) Architecture encapsulated in the URI syntax above willbe evident. DNS servers responsible for resolving the FRISB Realm mayauthoritatively delegate zones at any point in the “B dotted path” (thepath to the right of “@”), massively and automatically distributingservice over the geographic numbering plan.

FIG. 42 illustrates the dramatic distinction between a Level 2 (“zerozoned”) DNS resolution, and a Level N zoned and delegated DNS tree.

The upper panel wildcards all discs on the primary domain “frisb.com” byconfiguring the DNS A record to point all connections to a singleGateway D, represented by address IP(D), collapsing service to a singlecentral node.

The lower panel creates N zones to the left of the primary domain,configuring the DNS A record to point each zone N to its servicingGateway D(N), represented by address IP(DN), thereby distributingservice over N nodes.

Thus, whereas the upper panel would switch all calls to a single centralservicing gateway for routing logic, the lower panel automaticallyswitches connections to the distributed node N as configured by DNS.That is, the one and same URI yields different switching paths based onthe current DNS configuration.

By abstracting the routing logic to DNS, the service can “retroactivelyscale” by distributing the switching layer on creating more nodes in theDNS tree, all without modifying the programmed switching logic and URIassembly.

FIG. 43

Expanding on DNS zone delegation, FIG. 43 (F43) tabulates example DNS Arecords. The upper table distributes disc registration and switching bycountry, the lower table increases granularity by configuring DNS zonesdown to network or regional network operator.

With reference to the upper table, the first DNS A Record “wildcards”and catches all global disc registrations that are not zoned anddelegated to any particular region. The second DNS A record wildcardsall discs addressed to phone numbers beginning with US internationaldialing code “1” to point to the US located disc serve (IPx1) and so on.

The more granular service distribution is achieved on delegating zones“to the left along the dotted path” as illustrated by the examplerecords listed the lower table. Detailed zone delegation per the NANP,for example, is described in more detail with reference to FIG. 18above.

Continuing with playing FRISB . . .

In the case that a plurality of players spin discs to the same telephonenumber, even although, in one embodiment, discs from different sourcesmay present to the phone via the same generic call back number, simpleservice logic resident in either the disc browser or the Gateway mayeffectively route returns, as per the automatic answer and LIFO queuingmodels described earlier, in particular with reference to FIG. 33.

Notwithstanding the anonymous registration and generic callbackpresentation, the “loose coupling between A and B” nevertheless delivers“high cohesion”, since there is a 3rd (space) and 4th (time) dimensionto the game that is captured outside the service realm by the socialcontext. While Internet players A are anonymous to “the system”, theyare typically known to phone users B.

Given that FRISB is a “light communications product” with a high degreeof informality, connections are rendezvoused between friends, where forexample, either player may send the other an advanced game notificationusing any of a number of existing collateral communication channels(including email, instant text and even short telephone calls),establishing “the space” (who is on the net and fixed, who is mobile)and “the time” (to play).

For example, a typical rendezvous is where user A sends contact B amobile short message such as “frisb in 5 mins?” and on beingacknowledged, establishes the desired connection by spinning a disc to Bwho then “catches and returns the missed call to A” within theestablished window. Conversely B can invite Internet user A into thegame by sending a text message to telephone number A, requesting a disc(“spin me a disc in 5 mins?”).

Continuing with multiplayer FRISB . . .

By permitting multiple discs addressed to the same B phone player, theservice registers multiple “B net extensions”, as per the SOLO protocoldescribed above with reference to FIG. 33. As illustrated in the aboveembodiment, when B returns the gateway “group rings all the associatedextensions”. That is all the B registered discs ring “in parallel”,where the first disc to then “catch the call” and thereby answer andestablish communication, becomes the active “player”.

In this instance, when caller B returns, all such registered discs gogreen (ring), and the moment the first disc answers, all the remainingdiscs go orange (indicating a “missed call”). Since the connection isrendezvoused as previously described, the user A that addresses phonedestination B “last” (most recently) is typically the one to catch thereturn, since in all likelihood the prearranged call arrives within theautomatic answer window.

This “game theory” permits anonymous disc players “A1 A2 . . . An” tospin discs to a single phone player B and yet still establish thedesired connection without any additional address information and manualplayer selection on the return. Further, “multiplayer” engagementdelivers a powerful new way to conference call on the fly.

Unlike conventional “multiple endpoint registrations”, where a singleuser registers multiple VOIP phones at different locations (home, work,travel, mobile, fixed etc.) and where “ringing air” permits the singleowner to pick up at the preferred location, the multiple discregistrations disclosed here actually represent multiple players“clustered around a commonly registered phone identity”.

Continuing with the “ring all, first to answer wins” as described, oncephone user B has connected with player A1, B may place the active callon hold and redial the service gateway to establish a second connectionwith another player. Since player A1 is already connected to caller Band the endpoint (extension) is engaged, the second call from phone userB now rings all the remaining discs “A2 A3 . . . A(n)”.

In this call topology, the next disc player to pickup, “A2 then A3 andso on”, may thus be conferenced, on the legacy network side, into thealready established connection with player A1, using well understoodfixed and mobile network handset functionality. In such a multiplayergame, the phone acts as the “conference moderator”, selectively addingand removing discs (note, in the above examples for legibility, A1 isassumed to be the most recent player).

Phone player B can invite and rendezvous connections with friends aroundthe world who spin virtual discs “catch the return” and get placed intoconference without cost whatsoever. Once again given the “informal”nature of the service, any particular player who answers a call and isnot intended to participate in the desired conference, disconnects andlets the next disc ring unanswered.

Legacy conferencing serves to illustrate how “truly converged networks”,networks that connect tangentially as symbolized by the infinity “oo”symbol, rather than networks that overlap and intersect and where theresultant total sum is lcss than the individual parts, leverages servicecapability on one network without replicating the functionality on theanother. The conferencing functionality available on legacy telephonesand networks thus obviates the need to build out call bridgingfunctionality on the net.

In a further embodiment, disc anonymity may be supplemented with an “IPgeo tagging” service that automatically pinpoints the Country and/orCity associated with the browser IP address. By example, if player A1spins a disc from an IP address range allocated to USA CA state, thenthe registry may tag the disc with the region identified and announcethe disc “source” to the caller B using standard text to speech.

Given the social context to the game, announcing a Country and Cityplayer list (fbr example announcing players from “US CA dot”, “IL dot”,“ZA dot” and so on) is sufficient to indicate selection to the caller.Additional standard IVR interactivity may permit the caller to selectindividual, multiple and all players.

Clearly, the service website may offer additional personalizationfeatures, permitting players to personal their game. For example, theuser may enter a “handle” to identify the disc. Typically this handlewould be the user's first name, nickname or simply the first and lastinitials, as in “AK”, since discs tagged with just two and three letterswill have relevance to the sender and receiver. In this instance theGateway IVR may then announce a quick “AB” player list for callerselection.

Identifying players by their initials have the additional “interactive”advantage in that IVR announcements are crisp and caller interactionminimal. Announcing the player(s) prior to connecting the call back tobrowser delivers “secondary caller line identification” thatpersonalizes a shared gateway access telephone number. This containscaller “exposure” in that the caller may disconnect from the serviceprior to ringing the browser, allowing the caller to “exit the game”without offend ing a player.

Alternatively, the disc player may select a feature to personalize thedisc by voice, wherein the web service initiates a registration call tocapture a player “sound bite”, where by example, users announce andidentify themselves by speaking their name, which is then stored forplay back at the servicing gateway. Such personalization may also beautomatically engaged on sending a first disc from the connectingbrowser. The identifying sound bite may also be locally recorded on thebrowser device itself and sent to the service registry for play back atthe Gateway.

In further identification embodiments, Internet player lists may bepresented to the disc identified telephone using alternate bearers andprotocols. In one such embodiment, phones may receive an SMS messagelisting the players and the associated Gateway access number. Thispermits instant and “silent” selection, where the phone user may dialthe Gateway number embedded in the SMS and immediately key the initials,for example pressing “25” to select player “AK”, without having tolisten to IVR prompts.

In a related embodiment, the user may at any time ring the Gateway anddisconnect the call prior to answering, and thereby “ping the gateway”and request delivery of such an SMS player list at no cost. In yet afurther embodiment FRISB telephones identified by internet users mayregister their email address with the service and receive emailnotifications of players and pending games.

In a more tightly coupled version of the game, discs arrive on thedestination telephone presenting a Gateway access number that uniquelyassociates and identifies “internet player A to telephone number B”, asper the PICO protocol described. This “primary identification” permitsphone disc recipients to screen Internet invites by number.

Significantly, as the game revolves around the identified legacy phonenumber B and since the FRISB phone is embedded directly in the webbrowser page and is automatically loaded and registered when accessed,FRISB can present a “viral B” telephony link to global freecommunication.

In what can best be described as a “reverse disc spin”, where telephoneuser B, rather than net user A, invites users into the game, FRISBdelivers a compelling email signature line, as illustrated by thefollowing “tel net link” example:

“www.frisb.com/14154125111”

This “1click 2global free telecommunications” link instantly presentsthe browser embedded and automatically registered FRISB sessionprogrammatically preaddressed to the linked phone number, allowing anyinternet user to establish a frictionless global callback sessionwithout any specialized client software, setup and cost.

Since everyone knows their own telephone number, and anyone can sendsuch a text link “without permission”, the disclosed method and systemcould create the delivery of a pushed and virally propagated service,once again, without downloading software, without registering forservice and without paying to connect and talk.

While “call buttons” can mask the phone identity and may be easilygenerated using well known scripting methods (for example, entering aphone number in a web page that generates the associated andautomatically constructed web object), the value in displaying theactual URL in “clear text” as shown above, is it virally propagates theservice “forward with instruction”. That is, it shows recipients of aFRISB telephony link how to construct and send their own telephony linkonwards to others.

In a related embodiment, the FRISB architecture similarly extends gameplaying directly over the Internet between two browsers, in addition toplaying between Internet and phone. “FRISB on the net” is a uniquelyself discovering service that instantly matches two discs launched onthe same URL, connecting them peer to peer.

In this related embodiment, Internet users A1 and A2 share a “secretdisc handle” over a complimentary communications channel such as SMS,email or conventional phone call, to engage a new category of speechdubbed “Instant Voice”. By example, A1 and A2 share the alpha(numeric)disc handle “panther” launched on the following link:

www.frisb.com/panther

Using an adapted ENUM (“ANUM”) variant to cater for alphabetic servicename distribution, the FRISB server renders a realm that encodes thehandle according to the following algorithm:

1. IF the first disc letter is alpha then it constitutes the L3 domain(as with the “p” in “p.frisb.com” for “panther”)

2. IF the first disc letter is numeric then the handle is prefixed witha leading alpha character to prevent zone intersection with the legacyphone realm. For example, if the shared secret handle is “147ak”, thesystem registered handle would be “Z147AK”

3. Every subsequent letter is encoded using the ABC/2 alpha/numericalmapping schema as displayed on the conventional mobile keypad, todeliver a sufficiently distributed digital zoning schema.

4. The resultant domain is reversed as per the ENUM specification

Using the above, the resultant DNS tree has “A through Z” as the primaryL3 delegated zone, each followed by numeric L4 and greater zones as perthe ABC/2 encoded schema. Thus, a “PANTHER” titled disc (capitals foremphasis) would result in the following registered domain:

7.3.4.8.6.2.P.frisb.com

Where the ABC/2 encoding on the reversed dotted PANTHER string,excluding the first letter, is R.E.H.T.N.A.(P) yields the numericsequence:

pqRs/7. dEf/3. gHi/4. Tuv/8. mNo/6. Abc2. (P)

Describing the discovery process in more detail, user A1 enters the“www.frisb.com/panther” URL into a browser. FRISB returns with theembedded phone applet page as described above. Panther A1 goes red.

On successfully embedding the phone applet, the disc programmaticallyregisters itself on the ANUM URI realm, as above. Panther A1 goesorange.

On disc registration, disc A1 invites a connection, using the same URIit registered with, to locate its mated pair. During this invite, discA1 “engages its line”, that is it goes off hook rejecting any incomingcall, to prevent a loop back connection with itself.

Since A2 has not yet linked to “www.frisb.com/panther”, the initialinvite by A1 fails to locate its mated disc.

A2 now enters the “www.frisb.com/panther” URL into a browser and loadsas above. Panther A2 goes red.

On disc registration disc A2 now similarly invites a connection with“itself” in an attempt to locate its mated pair. During this invite,disc A2 similarly “engages its line” to prevent a loop back connection.

Since A1 is linked and registered as “panther”, A2 locates its matedpair.

Disc A1 instantly goes blue, automatically answering the call andskipping the ringing state green, in one embodiment.

Disc A2 similarly goes blue on detecting call answer by A1.

The connection establishment is as magnetic as it is redefining. Discs“pop into the blue” to deliver “instant voice” between two commonlylinked browsers. Metaphorically speaking, FRISB over the net delivers an“IP Intercom” on the fly.

If A1 or A2 inadvertently discovers and connects to an unknown A3opening the same disc handle at the same moment in time, “serendipityconnects two strangers”. If stronger disc channel privacy is required,users select longer and more personal disc handles.

The FRISB experience veils IP Telephony “systems complexity”, much asthe “on/off” light switch masks a billion dollar power grid deliveringelectricity to the mass consumer. By insulating the user from the“tangled wires and underlying circuitry”, from all the setup,configuration and the overwhelming arrays of non essential IP telephonyfeatures, FRISB allows Internet and telephone users to “connect andtalk”.

By converging technology and culture without, FRISB delivers a service,where the net essence delivers a core frictionless, and where thetelephone engages “virtually” at the pinnacle of the telephony realestate, inviting the phone user into the game via the “missed callpresentation manager”.

Operating as it does on the defining Internet link and exclusively on“the most pivotal key in the cellular universe”, FRISB is engaged with“1 link” on the web and “1 click” on the phone. User A clicks a link andB presses SEND (Green) in order to return the disc, to call the gateway,and establish the connection back to the net player. FRISB hooks intothe “mass click and dial stream”.

Further in leveraging existing mobile telephone technology and mobilecommerce on the reverse caller association and connection establishmentfrom “phone back to net”, FRISB and the disclosed methods and systems,inherit the mass established cellular billing model without coercingpayment on the net.

By preserving the “mass net free proposition” thus, playing digitalFRISB pulls the “mass mobile paying” into the game, potentiallygenerating additional inbound (interconnect) revenue for the InternetService Provider in terminating the call on net, rather than incurringthe outbound (terminating) penalty which would be the conventionalforward VOIP switching case.

The plurality of systems methods and protocols disclosed describevarying degrees of “coupling” between IP and Telephony domains.Individually and collectively the methods and systems yield a new classand order of IP switching matrices between conventional and NextGeneration Networks. The disclosed systems methods and protocolsdescribed above may be summarized as depicted in FIGS. 44 through 48.

FIG. 44

NANO. Number as named object.

NANO delivers a “1:many:1” (Telephone B to VOIP A to Telephone B)switching relationship. As disclosed, advance association with Bautomatically establishes a numbered VOIP client with contacts A. ManyVOIP users A(xyz) can be reached by one telephony user B, who converselymay be reached by many VOIP users, when “ON net” as disclosed.

This switching matrix is depicted in FIG. 44. Telephone user B dialsgateway C, where users A, who have previously identified B by number,are presented as “net contacts”. C instantiates VOIP client proxy B(#)that publishes “B online tele-presence indication” to users A,permitting them to terminate outgoing connections on net with B, via theproxy.

FIG. 45

ALTO. Authenticated line to owner.

ALTO delivers a “many:1” (Telephone B to VOIP A) switching relationship.As disclosed, VOIP client A registers an existing legacy phone number A#as net alias. Participating legacy network operators assign a symbolicnet prefix (#) that permits users to dial IP direct on the prefixednumber, thereby connecting to the net aliased client. Many phone users Bcan thus reach one VOIP user A, who has aliased their net identity.

This switching matrix is depicted in FIG. 45 (F45). Legacy telephoneusers B symbolically prefixes phone number A registered to VOIP user A.The originating switch routes the connection on the prefix to aservicing IP gateway, which maps the A number to the VOIP A useraddress, ringing the net client.

FIG. 46

UNNO: Universal net number.

UNNO delivers a “1:1” (Telephone B to VOIP A) switching relationship. Asdisclosed, a universal carrier independent Virtual Country prefixuniquely routes to registered VOIP user A resolved using ENUM NAPTRrecorded on a private or public domain.

This switching matrix is depicted in FIG. 46 (F46). Telephone user Bdials a triple digit country code prefix that in one embodiment resolveson Global Title Translation to an Internet Multimedia Subsystem (IMS)node on the originating network. IMS resolves the dialed digits via DNSto the appropriate URI.

FIG. 47

PICO: Paired index coupling.

PICO delivers a “many to many” (Telephone B to VOIP A) switchingrelationship. As disclosed, Gateway C automatically slot assigns eachVOIP user A dialing phone B a uniquely paired DID number from a sharedpool of Gateway numbers. Many VOIP users A can thus present a primarycalling line identity to B who can then directly dial many VOIP users A.

This switching matrix is depicted in FIG. 47 (F47). Telephone user Bdials gateway access number C(N), which indexes the B array to determinethe assigned VOIP user identity A. The number pairing is only uniquebetween A and B and typically just “10 base pairs” (C0 through C9) aresufficient to uniquely map all phones B with all their net contacts A.

FIG. 48

SOLO. Switched on line origination.

SOLO delivers a “1:many” (Telephone B to VOIP A) switching relationship.As disclosed, anonymous VOIP clients A enter phone number B to registeras “B net extension” on an ENUM realm. One telephony user B can thusreach many linked VOIP users A.

This switching matrix is depicted in FIG. 48 (F48). Telephone user Bdials gateway C, which formulates a SIP ENUM URI on the B callingidentity, that in one embodiment locates and group invites all netextensions A. Alternatively C invites the last registered extensionfirst.

The foregoing description, for purpose of explanation, has beendescribed with reference to specific embodiments. However, theillustrative discussions above are not intended to be exhaustive or tolimit the invention to the precise forms disclosed. Many modificationsand variations are possible in view of the above teachings. Theembodiments were chosen and described in order to best explain theprinciples of the invention and its practical applications, to therebyenable others skilled in the art to best utilize the invention andvarious embodiments with various modifications as are suited to theparticular use contemplated.

What is claimed is:
 1. A controller for authenticating a user on anInternet connected device using a telephony subscriber device of theuser, the telephony subscriber device having a corresponding telephonenumber, the controller configured for: (a) receiving the telephonenumber from the Internet connected device; (b) initiating a phone callvia a telephony gateway to the user's telephony subscriber device, thephone call presenting a calling line identity; (c) recording a telephonenumber authentication record, the telephone number authentication recordcomprising an association between the telephone number and the callingline identity; (d) receiving the calling line identity sent to thetelephony subscriber device; (e) retrieving the calling line identitystored in the associated telephone number authentication record via alookup using the telephone number; (f) authenticating the telephonenumber if the calling line identity received matches the calling lineidentity retrieved from the telephone authentication record associatedwith the telephone number or authenticating the telephone number if thecalling line identity received matches a series of randomly generateddigits displayed in the caller line identity as recorded in thetelephone number authentication record.
 2. A non-transitory computerreadable medium having a computer program stored thereon that can beloaded onto a server connected through a network to a client computer,so that the server running the computer program constitutes a controllerin a system according to claim
 1. 3. The controller of claim 1, whereinthe calling line identity comprises a series of randomly generateddigits.
 4. The controller of claim 1, wherein the calling line identityis received from the Internet connected device.
 5. The controller ofclaim 1, wherein the calling line identity is received from thetelephony subscriber device returning a missed call to the calling lineidentity displayed on the telephony subscriber device.
 6. The controllerof claim 1, wherein the controller is configured for authenticating thetelephone number determinant on the calling line identity received atthe gateway if the calling line identity received from the Internetdevice, matches either in full or partially, the calling line identitysent displayed at the telephony device.
 7. A method of authenticating auser on an Internet connected device using a telephony subscriber deviceof the user, the telephony subscriber device having a correspondingtelephone number, the method comprising the steps of: (a) receiving thetelephone number from the Internet connected device; (b) initiating aphone call via a telephony gateway to the user's telephony subscriberdevice, the phone call presenting a calling line identity; (c) recordinga telephone number authentication record, the telephone numberauthentication record comprising an association between the telephonenumber and the calling line identity; (d) receiving the calling lineidentity sent to the telephony subscriber device; (e) retrieving thecalling line identity stored in the associated telephone numberauthentication record via a lookup using the telephone number; (f)authenticating the telephone number if the calling line identityreceived matches the calling line identity retrieved from the telephoneauthentication record associated with the telephone number orauthenticating the telephone number if the calling line identityreceived matches a series of randomly generated digits displayed in thecaller line identity as recorded in the telephone number authenticationrecord.
 8. The method of claim 7, wherein the calling line identitycomprises a series of randomly generated digits.
 9. The method of claim7, wherein the calling line identity is received from the Internetconnected device.
 10. The method of claim 7, wherein the calling lineidentity is received from the telephony subscriber device returning amissed call to the calling line identity displayed on the telephonysubscriber device.
 11. The method of claim 7, comprising authenticatingthe telephone number if the calling line identity received from theInternet device, matches either in full or partially, the calling lineidentity sent to the telephony device.